June 2014 - sr-users - kamailio.org

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

Problem with t_relay() and VPN tunnel
by Carlos Ruiz Diaz 24 Jun '17

24 Jun '17

Hi, I have two SIP servers, *A* and *B*, connected each other though a OPENVPN tunnel. The server *B* needs to t_relay() every SIP message containing the method MESSAGE to the server *A* but these messages never reach destination. I have tested the tunnel connectivity and works fine. I wrote a Perl script (located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying to figure out what is happening but these messages are received by *A* and processed correctly but when *B* does the same from Kamailio, it is never received. Here is the route part of kamailio.cfg in *B*: Observation: ($rU == "1004") result is *true* * if(is_method("MESSAGE")) { if($rU == "1004") { xlog("L_INFO","En 1004"); rewritehost("10.8.0.1"); if (!t_relay()) xlog("L_INFO","MIO Error en t_relay"); t_reply("200", "Ok"); xlog("L_INFO","MIO despues rewrite"); exit; } .....* The perl script that WORKS: *$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2 From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796 To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>> Call-ID: 19722852989(a)192.168.1.2 CSeq: 15773 MESSAGE Contact: <sip:2002@**PublicIP-protected**:5060> Max-Forwards: 29 User-Agent: DBL Content-Type: text/plain Content-Length: 34 +595981[protected] hello from kamailio '; use IO::Socket; my $sock = IO::Socket::INET->new( Proto=>'udp', PeerHost=>'192.168.2.102', PeerPort=>'5060'); print "Sending msg $msg\n"; $sock->send($msg) or die "error sending $!\n"; *Please help! Thanks in advance. Carlos.

3 3

Cannot hear voice with symmetric NAT and STUN
by Khoa Pham 08 Feb '17

08 Feb '17

Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT, and RTP server is for symmetric NAT. Supposed A calls B. If A, B both use symmetric NAT and STUN, they cannot hear each other If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each other. Why is that? I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt for Prediction Failure, is that related to this problem ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

8 18

Using the auth_ephemeral module
by Hemanshu Vadehra 17 Jan '17

17 Jan '17

I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example? Regards, Hemanshu Vadehra hemanshu.v(a)directi.com

5 8

Record-Route header
by jignesh gandhi 01 Apr '16

01 Apr '16

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

2 4

Kamailio TLS
by Kethzer Docteur 12 Feb '16

12 Feb '16

Im trying to enable Kamailio TLS but the configuration is not running at all when I cahnge it liike this as follow. Kamailio 3.3 ubuntu 12.04 LTS #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif Please help -- Kethzer Docteur

4 3

Append to reply in default action of failure_route
by Efelin Novak 28 Dec '15

28 Dec '15

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

3 3

kamailio 4.0.1 websocket configuration
by Rupayan Dutta 28 Dec '15

28 Dec '15

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

4 3

Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

3 6

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

4 8

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

11 24

Kamailio: MI Command: 'trusted_dump' and 'trusted_reload'
by José Luis Millán 22 Apr '15

22 Apr '15

Hi, I'm using 'allow_trusted()' from Permissions module without a problem in a Kamailio 4.0 routing logic. It does just work, but the MI command 'trusted_dump' tells me that the module is not in use: ''' server:/home/jmillan# kamctl fifo trusted_dump 500 Trusted-module not in use ''' Doing a 'trusted_reload' does neither make any query to database to retrieve the data. Other MI commands of same module do work correctly. ie: 'address_reload' or 'address_dump' Note: I have tested in a Kamailio 1.5 and the result is the same. Regards. -- José Luis Millán

4 3

File transferring issue
by Wingsravi R 25 Mar '15

25 Mar '15

Dear Kamailio'ns, I am working on kamailio Server (V 4.0) installed on Ubuntu 12.04 system. Currently i am trying file transfer feature using IMSDroid SIP clients. file transferring is fine between two IMSDroid clients but at the receiving end (file recipient) i cont see the sender identity/ sender name. it is just displaying as 'null' (Screen shot is attached). But the full file is transferring to destination address without any problem. I have configured MSRP module related script in my kamailio configuration file (find the attachment). but i guess it is not invoking anyway when file transferring and the file transferring is negotiating client-to-client basis. And also find Sip capture of file transfer (ngrep based) for better understanding. What could be the problem? Why at the file recipient side it is not displaying sender's identity (instead it is 'null')? how can i resolve this issue? Anyone please help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

5 6

Fwd: Need a support for Siremis and kamailio server
by Gaurav Rai 14 Jan '15

14 Jan '15

Hi I am using a siremis and *kamailio * hosted server with lylix.net and they give a link for getting a support about Siremis. I need a support to how to make OUT call on PSTN with registered subscriber on Softphone. Currently we have registered user on subscriber table with their location and can make a PEER to PEER audio call and text messaging also. To make OUT call on PSTN we had create a LCR gateway and LCR rule with Prefix but still can not make a OUT call getting a error 404 can u please help me out to make a PSTN call Waiting for your reply Thanks Gaurav Rai +919716238804

2 1

CNXCC PrePaid module
by Rene Montilva 27 Nov '14

27 Nov '14

Hi list I don't understand prepaid logic calculation, because when my credit is 0 or lees than cost per second, the call is established equal $var(i_pulse) and $var(f_pulse) how this works????

3 16

Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

2 1

Hash Table capacity
by Ricardo Martinez 21 Oct '14

21 Oct '14

Hello list. I have a question about the HTABLE module. Suppose that I have a mysql database with 2 million of records. Is possible to load this table as a hash table compatible with Kamailio!? Hope someone can help me. Thanks, Ricardo Martinez.-

6 15

Re: [SR-Users] Crash Kamailio 4.1.4
by Igor Potjevlesch 10 Oct '14

10 Oct '14

Hello, We updated this morning Kamailio in 4.1.4 with your patch. Kamailio crashed again this afternoon. here an extract from /var/log/messages : Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2] ua=[(innovaphone IP6010/9.00 hotfix24 [9.061271/9061271/300])], INVITE from 'untrusted' host Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2], INVITE from an authorized SIP trunk Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2], INVITE from an authorized SDA for current SIP trunk Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri 0123456789 was rewritten to sip:0123456789@GW, carrier 1, domain 1 Jun 25 13:49:02 /usr/local/sbin/kamailio[20259]: : <core> [mem/q_malloc.c:140]: qm_debug_frag(): BUG: qm_*: fragm. 0x7f12803cb450 (address 0x7f12803cb480) beginning overwritten(abcdefed)! And this is the btfull : #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x0000000000546d3c in qm_debug_frag (qm=0x7f1280275010, f=0x7f12803cb450) at mem/q_malloc.c:142 #3 0x0000000000548b26 in qm_free (qm=0x7f1280275010, p=0x7f12803cb480, file=0x6276a0 "<core>: parser/parse_ppi_pai.c", func=0x627a00 "free_pai_ppi_body", line=102) at mem/q_malloc.c:464 #4 0x000000000056e5e6 in free_pai_ppi_body (pid_b=0x7f12803cb480) at parser/parse_ppi_pai.c:102 #5 0x000000000054fee0 in clean_hdr_field (hf=0x7f1274c3c238) at parser/hf.c:126 #6 0x00007f127cb6dde6 in acc_onreply (t=0x7f1274c157f0, req=0x7f1274c3ac08, reply=0x7f12804a6d70, code=200) at acc_logic.c:501 #7 0x00007f127cb6e30a in tmcb_func (t=0x7f1274c157f0, type=512, ps=0x7fff0b015580) at acc_logic.c:573 #8 0x00007f127ed68478 in run_trans_callbacks_internal (cb_lst=0x7f1274c15860, type=512, trans=0x7f1274c157f0, params=0x7fff0b015580) at t_hooks.c:290 #9 0x00007f127ed6868a in run_trans_callbacks_with_buf (type=512, rbuf=0x7f1274c158b0, req=0x7f1274c3ac08, repl=0x7f12804a6d70, flags=200) at t_hooks.c:336 #10 0x00007f127ed9ac06 in relay_reply (t=0x7f1274c157f0, p_msg=0x7f12804a6d70, branch=0, msg_status=200, cancel_data=0x7fff0b0158e0, do_put_on_wait=1) at t_reply.c:2001 #11 0x00007f127ed9d0b7 in reply_received (p_msg=0x7f12804a6d70) at t_reply.c:2499 #12 0x000000000045d837 in do_forward_reply (msg=0x7f12804a6d70, mode=0) at forward.c:777 #13 0x000000000045e0f8 in forward_reply (msg=0x7f12804a6d70) at forward.c:860 #14 0x00000000004a58e7 in receive_msg (buf=0x924600 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK512b.82b197888826f6b60c0c63b79801294d.0;received=A.B.C.D\r\nVia: SIP/2.0/UDP A.B.C.D:2057;branch=z9hG4bK-129F259C;rport=2057\r\nCall-ID: cb0"..., len=1124, rcv_info=0x7fff0b015c60) at receive.c:273 #15 0x000000000053c9a8 in udp_rcv_loop () at udp_server.c:536 #16 0x000000000046d42b in main_loop () at main.c:1617 #17 0x0000000000470533 in main (argc=7, argv=0x7fff0b015f98) at main.c:2545 It seems to be the same problem but in a different source. Can you help me? Regards, Igor. 2014-06-12 17:46 GMT+02:00 Igor Potjevlesch <igor.potjevlesch(a)gmail.com>: > Hello, > > we didn't set the early media parameter . its '0' by default, isn't it? > > regards, > > Igor > > > > > 2014-06-12 17:03 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, >> >> if you get a record for 180 response, then you have also the early_media >> parameter set for acc module, isn't it? >> >> In the morning I pushed a patch that should fix this issue. Use latest >> release 4.1.4 and see if works fine. Report the results to know that it is >> gone or not. >> >> Cheers, >> Daniel >> >> >> On 12/06/14 16:55, Igor Potjevlesch wrote: >> >> Hello, >> >> We don't use $ai in xlog nor in other process. only in ACC. >> >> parameters for ACC are : >> modparam("acc", "db_flag", FLT_ACC) >> modparam("acc", "db_missed_flag", 3) >> modparam("acc", "db_url", DBURLWO) >> modparam("acc", "db_extra", >> "src_user=$fU;username=$Au;src_domain=$fd;src_ip=$si;src_pai=$ai;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") >> >> For the 3781-4b1-572014182635-OGNAJ-1-A.B.C.D there is a code 180 >> ringing in the INVITE in ACC table. >> >> regards, >> >> Igor >> >> >> 2014-06-11 23:01 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Few more things... >>> >>> Are you recording 1xx events? Can you check to see if there is another >>> record in acc table for the same call? You can search by call-id >>> 3781-4b1-572014182635-OGNAJ-1-A.B.C.D >>> >>> Eventually, send also the parameters you set for acc module. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 11/06/14 19:25, Daniel-Constantin Mierla wrote: >>> >>> Hello, >>> >>> so you don't print $ai in xlog() statements nor use it in any >>> assignments or other functions besides acc parameter? >>> >>> Cheers, >>> Daniel >>> >>> On 11/06/14 19:19, Igor Potjevlesch wrote: >>> >>> Hello, >>> >>> We do not access to the P-asserted-identity in our configuration but >>> we added the field PAI in the db base ACC ( for INVITE, ACK and BYE) . >>> I dont know if it's in request_route, failure_route or branch_route . >>> >>> This is the print : >>> >>> (gdb) p mem_block >>> $3 = (struct qm_block *) 0x7f6a6bef1010 >>> (gdb) p shm_block >>> $4 = (struct qm_block *) 0x7f6a5666a000 >>> >>> Regards, >>> >>> Igor >>> >>> >>> 2014-06-11 18:02 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>> >>>> Hello, >>>> >>>> cloning to shm for tm seems ok. Can you tell where you access >>>> P-Asserted-Identity header, via variables? Does it happen in request_route, >>>> failure_route or branch_route? >>>> >>>> Can you print from gdb, any frame: >>>> >>>> p mem_block >>>> p shm_block >>>> >>>> I want to see if parsed filed point to shm or pkg memory. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 11/06/14 17:37, Daniel-Constantin Mierla wrote: >>>> >>>> Hello, >>>> >>>> at least I narrowed it down a bit. It is empty also in the clone stored >>>> in transaction, so it happens either during cloning or before. I will have >>>> to check these parts. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 11/06/14 17:00, Igor Potjevlesch wrote: >>>> >>>> Hello, >>>> >>>> This is the result, always for frame 5 : >>>> >>>> (gdb) p *t->uas.request->pai >>>> $1 = {type = HDR_PAI_T, name = { >>>> s = 0x7f6a60cd34b8 "P-Asserted-Identity: \"0987654321\" >>>> <sip:0987654321@A.B.C.D>\r\nContact: <sip:0987654321@A.B.C.D:5060>\r\nAllow: >>>> INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, >>>> REFER, UPD"..., len = 19}, body = { >>>> s = 0x7f6a60cd34cd "\"0987654321\"<sip:0987654321@A.B.C.D>\r\nContact: >>>> <sip:0987654321@A.B.C.D:5060>\r\nAllow: INVITE, BYE, REGISTER, ACK, >>>> OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE\r\nSupported: >>>> path,"..., len = 43}, len = 66, parsed = 0x7f6a6d81da88, next = >>>> 0x7f6a60cd3f10} >>>> >>>> (gdb) p *((p_id_body_t*)(t->uas.request->pai->parsed)) >>>> $2 = {id = 0x0, num_ids = 0, next = 0x0} >>>> >>>> *Did *you find one thing in common between the 2 occurrences? Do you >>>> have any ideas about what is the cause of this pai reset? >>>> >>>> Regards, >>>> >>>> Igor >>>> >>>> >>>> >>>> >>>> 2014-06-11 16:09 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com> >>>> : >>>> >>>>> Hello, >>>>> >>>>> in the same frame 5, can yo get: >>>>> >>>>> p *t->uas.request->pai >>>>> p *((p_id_body_t*)(t->uas.request->pai->parsed)) >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> On 10/06/14 18:35, Igor Potjevlesch wrote: >>>>> >>>>> Hello, >>>>> >>>>> Here is the results : >>>>> >>>>> (gdb) frame 5 >>>>> #5 0x00007f6a687e9b43 in acc_onreply (t=0x7f6a60d16ff8, >>>>> req=0x7f6a60cd2c10, reply=0x7f6a6c119aa8, code=200) at acc_logic.c:471 >>>>> 471 acc_db_request(req); >>>>> >>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>> >>>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

2 24

RFC: updating default values
by Daniel-Constantin Mierla 02 Oct '14

02 Oct '14

Hello, I am opening this discussion to decide if there is need to adjust some of the default values we have in Kamailio. Many of them were set even like 10 years ago, so they might not be very actual anymore. The main goal is to get the best possible settings for common usage. To start with, here are some values that should be reviewed: - default private memory is 4MB - if the config is not that small, it might not be sufficient free pkg to play with (e.g., for sql_query() result, storage of $var(...) values). Should it be increased to 8MB or other value? Debian/Centos have startup script that sets its value to 8 via command line - default shared memory is 32MB - for a decent deployment with tm, location, lcr/dispatcher, permissions, and anti-flood, it might leave not much free space. Should we double it or set to a different value - usrloc - db_ops_ruid should enabled (1) - seems stable, without it there are problems updating/deleting location records when UA changes the call-id for same contact address. - usrloc - hash_size - now is 9, which results in 512 slots, meaning is ok for few thousands of registered users, for more, performance will decrease when doing save/lookup location -- should it be made 10 (1024 slots for internal hash table)? - auth_db - load_credentials defaults to 'rpid', meaning that the query to get the password will retrieve also the rpid column. I haven't see rpid being used that much lately (replaced by PAI/PPI). I would make this parameter empty by default to avoid querying for an extra column that is likely to be empty. Perhaps there are more, I just wanted to get started. Reply with your comments to above list as well as add new items you thinks their default values should be adjusted. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 6

XMLRPC problem
by Kelvin Chua 18 Aug '14

18 Aug '14

i am currently having problems on xmlrpc i am using node-xmlrpc and it fails when htable.dump returns more than 1 row. i raised this issue on the node-xmlrpc group and this is the explanation i got. Each <value> node inside <param> is supposed to only have one child node, so this looks like an invalid response. The<value> should (probably, i don't know your use case) contain an <array> with <data> wrapped around the structs. Seehttp://xmlrpc.scripting.com/spec.html does this mean kamailio is not following standards? here is the working response: <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> </value> </param> </params> </methodResponse> here is the non-working response: <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> <struct><member><name>entry</name><value><int>11</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>3::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> </value> </param> </params> </methodResponse> Kelvin Chua

4 6

Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
by Olli Heiskanen 06 Aug '14

06 Aug '14

Hello, I've been experimenting with Kamailio with ws and sip clients and could need a hand in getting a call between those two to work. I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS 6.5 and a mediaproxy-ng running. I have clients wsclient(a)testers.com and gsclient(a)testers.com and I try to make call from wsclient to gsclient. The wsclient is a jssip client running on chrome and gsclient is a grandstream desk phone. My config file is the default one enhanced by online examples. I use a html5 <audio> element for the media streams, and configured my jssip phone to accept audio options like this: var options = { 'eventHandlers': eventHandlers, 'mediaConstraints': {'audio': true, 'video': false } }; sipUA.call(callto, options); I used the instructions from here: http://www.slideshare.net/crocodilertc/webrtc-websockets What I get is gsclient ringing, and as I answer there is no audio and call hangs up in a few seconds. I guess this is a SDP problem, something between Kamailio and Mediaproxy-ng but SDP is not my strong point so I'd appreciate advice. Question is where's my misconfiguration/problem? I would like to learn why this problem occurs and how to fix it rather than getting a solution right away, but please bear in mind I don't know much about SDP. In Kamailio log I see: kamailio[27059]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Error rewriting SDP kamailio[27058]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id kamailio[27057]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id Following are the INVITEs and 200 OKs from my SIP trace (1.1.1.1 is the ip of my Kamailio & mediaproxy-ng box and 2.2.2.2 is the public ip behind which both my clients are). The gsclient has port 5066. ****************************************************************************** U 2014/04/01 20:03:41.060009 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 2211. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/SAVPF 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:3350409123 1 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:3350409123 2 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0. a=candidate:4233069003 2 tcp 150995 T 2014/04/01 20:03:41.119806 2.2.2.2:38986 -> 1.1.1.1:5060 [A] ...... U 2014/04/01 20:03:41.159086 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Warning: 304 GS "Media type not available". Content-Length: 0. . U 2014/04/01 20:03:41.159392 1.1.1.1:5060 -> 2.2.2.2:5066 ACK sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Max-Forwards: 16. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 ACK. Content-Length: 0. . U 2014/04/01 20:03:41.161085 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 3136. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/AVP 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=fingerprint:sha-256 72:54:87:EC:D2:4C:D1:70:C2:FE:69:08:20:5C:92:1D:E0:EA:BD:45:09:E0:90:62:27:B6:34:60:54:E2:99:28. a=setup:actpass. a=mid:audio. a=sendrecv. a=rtpmap:111 opus/48000/2. a=fmtp:111 minptime=10. a=rtpmap:103 ISAC/16000. a=rtpmap:104 ISAC/32000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:106 CN/32000. a=rtpmap:105 CN/16000. a=rtpmap:13 CN/8000. a=rtpmap:126 telephone-event/8000. a=maxptime:60. a=ssrc:3298511848 cnam And here are the 200 OK messages when answering the call: U 2014/04/01 20:03:46.049711 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. T 2014/04/01 20:03:46.051127 1.1.1.1:5060 -> 2.2.2.2:38986 [AP] .~.dSIP/2.0 200 OK. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. ****************************************************************************** cheers, Olli

4 26

Sample Kamailio Config for Siremis
by Noah Mehl 02 Aug '14

02 Aug '14

I have managed to install Kamailio and Siremis 4.x, but I have noticed that neither Kamaliio or Siremis come with a sample config that makes sense for Siremis (as far as I can tell). For instance, if I want to use LCR with dispatchers, the sample Kamilio configs don’t have the dispatcher module loaded or configured. Am I missing something? Can someone please share theirs? Thanks! ~Noah

3 2

Warning message on SIP websocket
by Alex Villacís Lasso 29 Jul '14

29 Jul '14

Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Via: SIP/2.0/TLS 192.168.3.2:58657#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: Er/CgVyBNzVGREBmt+w3FTff8s4=#015#012Server: kamailio (4.1.4 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012>

2 1

Plz help: MSILO on TLS + cert based client auth enabled server
by Marc M. 29 Jul '14

29 Jul '14

Hi, I have a SIP server with TLS + client certificate based authentication. The server is listening for SIP(UDP) on port 5060, and for SIPS(TLS) on port 5061) The client certificate based authentication is enforced on port 5061. I would like to fire up MSILO and face now a strange problem: At each REGISTER, the stored messages get dumped. Dumped messages get sent to the clients from senders logged in over port 5060(UDP) as SIP:xxxxxxxxx@xx.xx.xx.xx however Dumped messages are NOT sent to clients from senders logged in on port 5061(TLS) as SIPS:xxxxxxxx@xx.xx.xx.xx It looks like the messages get dumped so M-dump() is working, but they will never enter the route. I would assume TLS is stopping the dumped messages to entering the route logic. I would either 1. somehow force m_dump() to pipe the messages over SIP(and not SIPS), or 2. disable client auth for messages recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc

2 1

Only a redirect server
by Jack Smith 29 Jul '14

29 Jul '14

Dear friends, Just new to this mailing list, new to Kamailio, and new to running a sip server. I would like to try only a redirect server. I had a look at http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg which was friendly, but then the default file kamailio.cfg looked frightening ! Could someone help me set up a minimal Kamailio installation only to handle state-less redirect? Best, Jack

4 4

Many instances of "new URI shorter than old URI" in log
by Alex Villacís Lasso 15 Jul '14

15 Jul '14

Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:49 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI I am not really sure of what this message is supposed to indicate. It looks something related to uac_restore_[from,to] functions, but some debugging statements just before the use show that the message is not originating from there. This message is showing up hundreds of times per day.

2 4

Would like to upgrade to 4.1.4+wheezy -- when will this be available?
by Will Ferrer 11 Jul '14

11 Jul '14

Hi We would like to upgrade or debian instances to the apt repo for 4.1.4+wheezy. Is this release soon to be stable and ready for use? We would like to upgrade so we can get around the memory leak that arises when many reloads occur as described here: http://sip-router.org/tracker/index.php?do=details&task_id=436 Thank you very much for any advice any one can offer us. All the best. Will Ferrer

2 3

MSILO - dumped messages get duplicated
by Marc M. 06 Jul '14

06 Jul '14

Hi, I have a Kamailio 3.2 setup up and running. Recently I configured msilo for offline message store. Essentialy the setup runs fine. Messages to users without location info getting stored. In case the user has location info, but cannot be reached, I made failover_route, so the messages getting stored in this case too. Messages getting dumped at the next REGISTER. However, I have a group of new mobile clients, with unstable connection. It happens many times that they send a REGISTER, but the connection brakes afterwards. It this case the message is getting dumped, but since it cannot be delivered, it runs on the failover_route which stores the message again. However, the previous message is not removed from the db, so the same message is now twice in the db. Next time the message either gets delivered in double, or in case of a new failure, it gets stored 2x2= four times. I need to avoid this duplicates but still make sure the message get delivered sooner or later. Any help is highly appreciated. Here is my routing table: thanks Marc ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } if (is_method("MESSAGE")) { log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n"); route(MSILO_MESSAGE); } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ##storing offline messages route[MSILO_MESSAGE]{ if (lookup("location")){ t_on_failure("FAIL_MESSAGE"); t_relay(); xlog("L_NOTICE","---------------------------------MI: ($mi), MB: ($mb)\n"); }else{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, storing offline message ID: ($ru)\n"); m_store("$ru"); sl_send_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, stored offline message ID: ($ru)\n"); }; exit; }; failure_route[FAIL_MESSAGE]{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, storing offline message ID: ($ou)\n"); m_store("$ou"); t_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, stored offline message ID: ($ou)\n"); exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); # MSILO - dumping user's offline messages m_dump(); log(1,"-------------------------MESSAGE PROCESSING: dumping offline messages\n"); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

3 4

RTP to SRTP encryption with Kamailio.
by Dani Kamailio 04 Jul '14

04 Jul '14

Hello, I need to build a VoIP system who receives SIP and RTP traffic in a public IP and encrypt both of them with TLS and SRTP respectively. The main point is to have security inside of the local network (I know it may sound unuseful). So, I was trying to build the whole system in Kamailio but I got stucked with the RTP to SRTP bridge and I do not really know how to do it. I know that there are some modules like rtpproxy-ng and rtpengine as media relay that can handle SRTP... any ideas? I just want to make sure that I am in the right way. If it could be done in Kamailio, have I to write the code to encrypt RTP, like the algorithm or something? Would it be convenient to send the RTP packets to Asterisk? (and Asterisk would somehow encrypt them?) I am pretty new with all of this, any help will be apreciate, thank you.

2 2

FW: Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 04 Jul '14

04 Jul '14

Hi, We are setting up a Kamailio Proxy with the Topoh module in place masking everything including CALL-ID. It seems to be masking everything just fine except for CALL-ID in BYE requests (verified with PCAPs) when received from downstream. Upstream works fine. Looking at the source code for the module 4.1.2 we see the unmasking being done like this: if(dialog) { /* dialog request */ th_unmask_ruri(&msg); th_unmask_route(&msg); th_unmask_refer_to(&msg); if(direction==1) { th_unmask_callid(&msg); } } It only seems to be unmasking for one direction. From my understanding BYE can be initated from both directions? Might be something I don’t understand here how the source code works but I wanted to ask for any pointers of help here.. been troubleshooting this for a good while now. Kind regards, Samuel

3 2

How to check message queue (buffer) in Kamailio
by AliReza Khoshgoftar Monfared 03 Jul '14

03 Jul '14

Hi, I had another simple question: In a kamailio server (proxy), how do I check the number of messages currently waiting for processing? Is there a variable that I can monitor, say, if I want to make a routing decision in my config based on the number of messages in the queue? Also, is it possible to get a head count by method? or is it only possible after fully parsing the message? I see that "ratelimit" module uses similar information, but I am not sure how to get the status of these queues that the module uses. Thanks

3 5

Replys though onsend_route?
by Jöran Vinzens 03 Jul '14

03 Jul '14

Hi all, is there a possibility to pass reply and failure though onsend_route as well? We try to capture via siptrace the incoming and outgoing packet. So far this is only possible with Requests. I've found this: http://lists.sip-router.org/pipermail/sr-users/2012-May/073224.html Can anybody confirm this is still the behavior? Does anybody have a solution passing reply though onsend_route or any other way to capture the outgoing reply via siptrace? Thanks a lot for your help. BR Jöran Vinzens -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 3

Making RLS presence work with Blink and Kamailio 4.1.4
by Alex Villacís Lasso 02 Jul '14

02 Jul '14

I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice calls work correctly. With presence, I have managed to make XCAP storing and retrieval work, in the sense that a document that is being stored with PUT can be later retrieved with GET on the same URL. However, I cannot make Blink use this support in order to actually monitor presence. As far as I understand, when user A subscribes to the presence state of user B, and user B is registered, user B is supposed to receive a notification that user A requests authorization for presence subscription. However, this is not working so far when both A and B use Blink. From what I can see through a packet capture, neither adding a new contact on the Blink window or having existing contacts result in any SUBSCRIBE packets for the contact users being sent to the Kamailio instance. If the subscription is meant to be performed some other way, I am not seeing it being done either. Of course, this results in presence changes not being exchanged between contacts. The only SUBSCRIBE packet I see from Blink is the one for the SIP account used for Blink. With MSRP, I have copied the example from the msrp.so documentation, but I have replaced the authentication shown with a static password, with an actual query using auth_db.so, the very same query used to authenticate other SIP packets. Through the packet capture, when user A tries starting a MSRP chat with user B, an INVITE is sent from user A to Kamailio, which has a special check to NOT send this invite to Asterisk, but rather route it itself, and it gets "correctly" routed to user B. Then user B attempts to start a MSRP session with Kamailio (not user A) without authentication credentials, Kamailio challenges user B for the credentials, and... that's it. User B makes no attempt to repeat the MSRP session with the required credentials, and the communication eventually times out. The presence failure might have something to do with the following exchange I see in the packet capture: GET /xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index HTTP/1.1 Accept-Encoding: identity Authorization: Digest username="avillacisIM", realm="pbx.elastix.com", nonce="U6xFz1OsRKNuU11dgljLCgGppv5lK6Eu", uri="/xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index", response="3dfdf556c9812f4bebb8b0cc82578ef9", algorithm="MD5" Host: pbx.elastix.com:5060 Connection: close Accept: application/pidf+xml User-Agent: Blink 0.8.2 (Linux) HTTP/1.1 404 Not found Via: SIP/2.0/TCP 192.168.3.2:55567 Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 That is, the Blink client tries to fetch the pidf-manipulation document, and Kamailio answers with a 404. In other cases Blink has responded with a PUT of a minimal document at the required URI (which gets stored to the database), but in this case (pidf-manipulation) this is not done. I am not sure of what is supposed to happen here. This 404 is the same regardless of whether there are other (Blink or otherwise) SIP clients connected. I think this has been asked before, but is there a working implementation of MSRP+presence for Kamailio that works with Blink? Or failing that, a complete kamailio.cfg that has been shown to work with at least one MSRP implementation?

2 5

TCP rest by peer
by Rob Moore 02 Jul '14

02 Jul '14

Hi All, Since we've been using sipPhones as a proxy for registrations and calls to phones over TCP. I've been seeing a couple of these crop up in the logs. /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:1131]: ERROR: tcp_read_req: error reading /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) We haven't had any complaints of problems with the phones so if these are not a problem I'd like to know how to clean them up. Any suggestions on cause or solution would be fantastic. Thanks in advance. Rob Kind regards, Robert Moore Telephony System Architect [Description: Aeriandi] m. +44 (0)7766 838 040 t. +44 (0)845 108 0308 [cid:image002.png@01CF8C82.823C49E0]<http://www.retailweektechnologyawards.com/winners-2014> Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info(a)aeriandi.com<mailto:info@aeriandi.com> / aeriandi.com<http://www.aeriandi.com/> This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).

2 3

msilo possible bug
by Kelvin Chua 02 Jul '14

02 Jul '14

Hi guys, i think i have stumbled on an msilo bug. m_dump() is not working. kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 so here we can see there is a message to be delivered. i turned on debug mode, then the intended destination of message logs in. then this is what i see on logs DEBUG: msilo [msilo.c:1082]: m_dump(): dumping [1] messages for <sip:XX85X80X74@1X8.1X9.1X7.1X3>!!! DEBUG: msilo [ms_msg_list.c:169]: msg_list_check_msg(): checking msgid=405 DEBUG: msilo [ms_msg_list.c:207]: msg_list_check_msg(): msg already in sent list. DEBUG: msilo [msilo.c:1089]: m_dump(): message[0] mid=405 already sent. using ngrep, no message was ever seen on the wire. then, kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 still the same. anybody experienced this? Kelvin Chua

2 2

installation siremis error
by Hossein Aghaie 01 Jul '14

01 Jul '14

hi after install kamailio 4.1.0 on ubuntu server,I try to install siremis-4.1.0 of this Doc : http://kb.asipto.com/siremis:install41x:main but to end setup,show this error : http://paste.fedoraproject.org/114029/ how to solved this problem? tnx

3 5

Kamailio Development Workshop, July 9-10, 2014, Paris, France
by Daniel-Constantin Mierla 01 Jul '14

01 Jul '14

Hello, I want to announce the next Kamailio Development Workshop, to take place in Paris, France, during July 9-10, 2014. This event is the next in its series targeting to show the internals of Kamailio and enable more people to become developers as well as let users of Kamailio to get more knowledge about the design of the application which can have relevant impact in operating deployment. Be aware that it is not a workshop about Kamailio installation and administration. Its content is about writing C code to extend Kamailio. More details can be found at: - http://www.kamailio.org/w/2014/06/kamailio-development-workshop-july-9-10-2… Similar to the past events, we are considering to have a social networking event in the evening of July 9 (most probably a dinner meetup at a nice place in Paris), which is open for everyone, each participant taking care of own expenses. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

how to compile module db_oracle on Centos 6.5
by Konstantin Khatskevich 01 Jul '14

01 Jul '14

Hello! 5. Installation Because it dependes on an external library, the oracle module is not compiled and installed by default. You can use one of the next options. * - edit the "Makefile" and remove "db_oracle" from "excluded_modules" list. Then follow the standard procedure to install Kamailio: "make all; make install". * - from command line use: 'make all include_modules="db_oracle"; make install include_modules="db_oracle"'. But "db_oracle: requires non-free Oracle client SDK" (http://lists.sip-router.org/pipermail/sr-dev/2013-October/021803.html) What can I do? -- Best regards, Konstantin Khatskevich

2 2

Child Processes Stopped kamailio 4.0.6
by Jonathan Hunter 30 Jun '14

30 Jun '14

Hi Guys, Sorry thought I better raise this in new stream to avoid confusion. We have another crash today, not sure if the below is of much use without debug? Thanks Jon Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18533]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18537]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18534]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18535]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18538]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18530]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18539]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18536]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 kamailio[2076]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2075]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2074]: ERROR: <script>: Routing Call To Asterisk serverJun 26 09:38:09 pbx kamailio[2074]: INFO: <script>: Routing Call to Dispatcher group 1 -Asterisk Media ServersJun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:788]: handle_sigs(): child process 2075 exited by a signal 11Jun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:791]: handle_sigs(): core was generatedJun 26 09:38:10 pbx kamailio[2072]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:10 pbx kamailio[2074]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2078]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2076]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2077]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2079]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2073]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2080]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2081]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2082]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2072]: ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: No such file or directory (2)Jun 26 09:43:33 pbx kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected)

2 2

How to extract information from location table
by aawaise 30 Jun '14

30 Jun '14

I want to know is there any indirect method to extract information from location table that is kept by Usrloc module ?? I want to know about the username and domain name of users from Location Table on arrival of INVITE packet. How can it be done ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-extract-information-from-loc… Sent from the Users mailing list archive at Nabble.com.

2 1

route [Location] Documentation
by aawaise 30 Jun '14

30 Jun '14

I want to know that how route [Location] works when an INVITE packet approaches kamailio server. Basically route [Location] gathers information whether callee is a valid user or not. Secondly if it is online or not. I want to understand this whole procedure. Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/route-Location-Documentation-tp1286… Sent from the Users mailing list archive at Nabble.com.

2 1

Voice quality issues when registering to Kamailio
by Travis Dillon 30 Jun '14

30 Jun '14

Hello all, I have an environment setup as follows: VoIP routers -> Kamailio 4.0 -> Asterisk 11.6 When I register the phones to Asterisk the call quality is excellent. So to be clear, the call comes in the VoIP router, routes through Kamailio to the phone registered to Asterisk. When I register the phone to Kamailio the call quality is significantly reduced. So clarity here, the call comes in the VoIP router, routes through Kamailio to the phone registered to Kamailio. Why would the call quality diminish when registered to Kamailio? Where can I start looking for this issue? Thank you, Travis R. Dillon

2 1

UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 30 Jun '14

30 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

4 4

Strange application hang querying DB
by Tim Chubb 27 Jun '14

27 Jun '14

Hi, I'm currently wracking my brains trying to deduce where an application hang can be coming from. I have ported the PostGres schema to be compatible with MS Sql Server and have had no issues getting a working dev environment working. However when I try and use SQL Azure as the backend db as opposed to SQL Server 2012 (running locally on my dev machine) Kamailio will seem to hang after about 10 - 15 mins of flawless operation. I have recreated the SQL Azure schema from the working schema running on my dev machine and copied all data in a 1:1 manner so I can be certain it's not a data quality issue. I am using the same FreeTDS/UnixODBC config that I am using for Asterisk realtime (which doesn't exhibit this behaviour). The really strange thing is that I can see the queries being sent by Kmailio in the freetds trace log, and the response from SQL Azure, however its as if the db response handler is deadlocked/hung. The kamailio process continues to work fine, and as long as its not required to commiunicate with the DB all appears well i.e. no error messages in the log file. This behaviour only happens when im using SQL Azure as opposed to SQL Server, so there must be something causing the issue, that's unique to SQL azure as opposed to SQL Server. As far as I can tell this is not something I can create a backtrace for as no core dumps are produced, as far as I can tell there is no easy way debug what the lock is. If anyone has any suggestions I would be very grateful to hear them. One thing that occurs to me is that the db_unixodbc module doesn't have half as many parameters as its equilent in Asterisk, and that connection pooling may be in use, this could be a bit of a show stopper AFAIK you have to disable connection pooling with SQL server, certainly any of the guides I have come across configuring Asterisk to work with SQL server suggests that. Tim Chubb Developer [VS-logo-email] The Coach House, Heywood House, Park Lane, Heywood, Westbury, BA13 4NA Direct: +44 (0)333 0110164 Email: tim.chubb(a)voicesimplified.com<mailto:tim.chubb@voicesimplified.com> Web: www.VoiceSimplified.com<http://www.voicesimplified.com/> Voice Simplified Ltd (registered in England & Wales: 07171825) registered office: Curzon House, 2nd Floor, 24 High Street, Banstead, Surrey. SM7 2LJ. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you are not the intended recipient, please return the message to the sender and delete it from your records.

2 3

Error when install Kamailio 4.1.4 on Mac OS 10.9.
by Son Truong 27 Jun '14

27 Jun '14

Hi Kamailio, - I meet an error when I am trying install Kamailio 4.1.4 on Mac OS 10.9. The error: touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 1: syntax error: unexpected end of file make: *** [install-cfg] Error 2 How can I fix it? I am new in Kamailio, I want to learn Kamailio. -- Thanks and best regards, Phạm Sơn Trường Android, iOS Team Lead at ME Corp. Email1: truonguit2010(a)gmail.com Email2: truongps(a)mecorp.vn Mobile: 01218816208

1 0

Processing of Invite Packet
by aawaise 27 Jun '14

27 Jun '14

When an invite packet is recieved by kamailio server, what processing is done on it ? Which modules and functions are called in its processing and in which sequence ? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Processing-of-Invite-Packet-tp12867… Sent from the Users mailing list archive at Nabble.com.

1 0

Transaction Stateful Message Handling
by AliReza Khoshgoftar Monfared 26 Jun '14

26 Jun '14

Hi, I am trying to configure a SIP proxy architecture that is stateful with regard to transaction but not the dialogues, i.e. If an INVITE takes a specific route, the corresponding 180 Ringing, 200 OK and ACK shall traverse the same set of proxies, but the corresponding BYE can take another route (and obv it's own 200 OK takes the same route again) Which modules can help me with that? In specific which functions / modules are recommended to handle requests and responses? Currently, I use "t_relay()" from TM module to forward requests and "loose_route()" from RR module to handle replies, but it looks like that if a BYE ends in a proxy which its corresponding INVITE has not traversed, it is not forwarded by that proxy. Am I making a mistake in my routing? Isn't there a simple way to do transaction stateful routing in Kamailio? Thanks, Alireza

1 0

How to stop kamailio from send 408?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, when t_relay() times out, calling sl_reply_error() will send 408 to the client. But if I don't want to send 408, how do I interpret the error code t_relay() set? Regards, Allen

2 2

How to find whether or not certain header exists?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, If I normally receive a INVITE that has a header 'aaa' with value 'bbb', but it's possible that some INVITEs don't have this header. How do I determine in the config file if the header exists? To check the value of the header I do if($hdr(aaa) == "bbb"). But what if the header doesn't exist? Cheer, Allen

2 2

Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 26 Jun '14

26 Jun '14

Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel's tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos

3 4

Trouble with rtpproxy
by Igor Potjevlesch 26 Jun '14

26 Jun '14

Hello, For a particular gateway, I force the RTPProxy. This work fine but on the other hand, in my side the RTPPRoxy response to the source IP address of SIP packet instead of the IP address in "Connection information". Is there a way to instruct the use of the IP address announce in the SDP instead of the last IP source of the last SIP packet? Regards, Igor.

2 3

segfault at 18 error 4 in auth_db.so on Version 4.0.6-1 Release
by Jonathan Hunter 26 Jun '14

26 Jun '14

Hi Guys, We had our kamailio instance crash today, it appeared to be after multiple registration failures, the resulting logs showing; Jun 25 15:47:55 kernel: : kamailio[14993]: segfault at 18 ip 00007f03102f9609 sp 00007fff1b346500 error 4 in auth_db.so[7f03102f1000+b000]Jun 25 15:47:55 /usr/sbin/kamailio[14989]: ALERT: <core> [main.c:788]: handle_sigs(): child process 14993 exited by a signal 11Jun 25 15:47:55 /usr/sbin/kamailio[14989]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 25 15:47:55 /usr/sbin/kamailio[14989]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLD Any comments welcome. thanks Jon

2 2

Syntax help in comparison of values of variables
by aawaise 26 Jun '14

26 Jun '14

I need some help regarding syntax. Can we put some comparisons on the values of these variables like if ("$ru"=="$fu") {xlog("both are same")}; If this is not the way then what is the way to do so ?? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Syntax-help-in-comparison-of-values… Sent from the Users mailing list archive at Nabble.com.

2 1

Extract information from invite packet
by aawaise 26 Jun '14

26 Jun '14

I want to log some information of callee from incoming invite packet. how can it be done. I want to get it logged in /var/log/messages. Any help will be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-information-from-invite-pac… Sent from the Users mailing list archive at Nabble.com.

2 2

Passing Command-Line Arguments to Kamailio Config Script
by AliReza Khoshgoftar Monfared 25 Jun '14

25 Jun '14

Hi, I had a simple question to which I could not find a clear answer in the documentation. How can I pass a command line argument to Kamailio config script? For example, assume that I run my kamailio daemon using script myconf.cfg: >>kamailio -f myconf.cfg Now I want to pass a value, say an IP address and use it inside the script. Sth like >>kamailio -f myconf.cfg --ip __MY_IP__ and then have an assignment of "$var=__MY_IP__" inside myconf.cfg Is there an easy way to do this?

2 2

Re: [SR-Users] IMS and SIP over WebSocket 500/Error
by DURECU, OLIVIER (OLIVIER) 25 Jun '14

25 Jun '14

Hi Carsten, I had a look Kamailio repo and did not find the configuration you proposed to post. Did I look at the bad place or didn't you find the time to do it ? Thanks, Olivier Hi Olivier, i will update the configuration files for WebSockets at the P-CSCF after KamailioWorld. The week after KamailioWorld, we are at another congress, but probably around easter, you should find a proper configuration for WebSocket on the P-CSCF (including media). Kind regards, Carsten 2014-04-01 12:02 GMT+02:00 DURECU, OLIVIER (OLIVIER) <olivier.durecu at alcatel-lucent.com>: > Hello all, > > I have installed Kamailio 4.1.2 in IMS configuration (3 Kamailio instances > to run in P/I/S-CSCF modes) > > Everything works fine with Native IMS clients (Boghe) > > I want now to introduce Web clients using WebSockets. I have added to the > P-CSCF Kamailio.cfg the websockets parts > Registration is Ok but I can't manage placing a call. > > With SIPoWS, the contact header is sip:olivier at 132.invalid. So when the > P-CSCF receives a terminating INVITE, its destination URI is > sip:olivier at 132.invalid , it tries to make a DNS resolution of 312.invalid > and it doesn't forward to the client saying the domain is unknown and can't > be routed > > I guess that something is missing in my configuration around > aliases/nathelper/outbound but didn't find what... > > Thanks for any help > > > Olivier DURECU > Bell Labs / IP Platform Research > Villarceaux Center > Tel: +33(0) 130772759 > GSM: +33(0) 683776482 > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 1

Setting priority (q) for kamctl ul add
by David Wilson 25 Jun '14

25 Jun '14

Thanks Daniel, Rather than changing the kamctl script, I can just run a command like: kamctl mi ul_add location <user> <uri> <expires> <q> 0 0 0 0 When I run this and set expires to 0 (as per the comment in kamctl, "expires 0 means persistent contact"), kamctl ul show shows expires=-1403662437 which siremis shows as 1970-01-01 08:00:00. Within 20 minutes, the entry is gone and I'm not sure why. Is the 8:00 (my timezone is UTC+8) causing a check for zero to fail or is there something else I'm missing? For reference, I'm using db_mode=2 for usrloc. Regards, Dave. On 20 Jun 2014, at 6:00 pm, sr-users-request(a)lists.sip-router.org wrote: > Date: Fri, 20 Jun 2014 07:38:50 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Setting priority (q) for kamctl ul add > > Hello, > > On 20/06/14 06:16, David Wilson wrote: >> Hi All, >> >> I'm trying to add a permanent usrloc entry via kamctl ul add. >> >> This works, but the created entry has a q value of 1.0 which is higher than I need. >> >> Is there a way to either: >> >> 1. Specify a q value when using kamctl ul add, or > apparently the Q is hardcoded, a way to fix it is to edit kamctl (which > is a shell script), search for 'ul_add' and update the 1.0 value in that > command. > >> 2. Edit the q value of an existing record by using a kamctl command. >> > Maybe removing and add it it with same other attributes... > > Cheers, > Daniel

2 2

Adding/manipulating headers in CANCEL
by Örn Arnarson 24 Jun '14

24 Jun '14

Hello, I'm running Kamailio 4.0.4, and I'm running into problems when trying to insert or update headers when the method is CANCEL. It just seems that nothing at all happens to the message. Is this a bug or a design choice? Regards, Örn

2 4

How to route on whether INVITE is for MSRP
by Alex Villacís Lasso 24 Jun '14

24 Jun '14

So far, I have a kamailio.cfg that routes INVITEs between Asterisk and the external networks. Now I want to start adding MSRP support. However, with my current configuration, the test client (Blink) sends an INVITE with a SDP payload that specifies msrps media. This gets routed to Asterisk, and it gets promptly rejected with "Not Acceptable Here". I need to make Kamailio route the request by itself if the INVITE is for MSRP, and delegate to Asterisk for other INVITEs. What is the most elegant way to test for the INVITE packet to see if it is MSRP? This testing should be at least mentioned in the documentation for the msrp.so module.

2 1

Setting up two Kamailio Proxies in Tandem
by AliReza Khoshgoftar Monfared 23 Jun '14

23 Jun '14

Hi, I am trying to build a simple architecture like this: UAC <--> Proxy1 <--> Proxy2 <--> UAS (each on a different VM) Is it possible to have this architecture built without any "registrar" module? I have simulated my UAC and UAS using SIPp and the routing in Proxy1 and Proxy2 is to forward to the next hop. for example in Proxy1, I have: *route{ record_route(); if (loose_route()) { route(FORWARD); break; } if (uri == myself){ setdsturi("sip:__ADRESS_OF_PROXY2__"); route(FORWARD);* * break;* * } route(FORWARD); }route[FORWARD]{ if (!t_relay()) { sl_reply_error(); }}* I simulate packets in UAC in the following manner: *sipp -sn uac -rsa __ADDRESS_OF_PROXY1__ __ADDRESS_OF_PROXY1__* The problem is that my first proxy (Proxy1) is always returning "404 Not Found" on any INVITE and never forwards to Proxy2, even though I have asked it to do so in "if(uri==myself)". Am I missing something in connecting the proxies in such a simple architecture?

1 0

kamailio not sending 202 after 408 is sent
by Allen Zhang 23 Jun '14

23 Jun '14

Hi all, I have an edge proxy (E) and a registrar (R) behind it. In R, if a MESSAGE failed to deliver for any reason, R stores the MESSAGE in msilo. If the MESSAGE timed out, R sends a 408 time out first and then send a 202 Accepted after the MESSAGE is stored. The problem is, E happily forwards the 408 to the sender of the MESSAGE, but decides not to forward the 202. The log: DEBUG: t_reply_matching: hash 16662 label 0 branch 0 DEBUG: tm [t_lookup.c:1032]: DEBUG: t_reply_matching: reply matched (T=0x7f99eebb2ae0)! DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=3 global id=3 T end=0x7f99eebb2ae0 DEBUG: tm [t_reply.c:2085]: DEBUG: reply_received: org. status uas=408, uac[0]=408 local=0 is_invite=0) DEBUG: tm [t_reply.c:1207]: ->>>>>>>>> T_code=408, new_code=202 DEBUG: tm [t_reply.c:1706]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Can I change this behaviour to also forward the 202 withouting hacking in the source code? Regards, Allen

2 4

Specifying Upstream and Downstream Servers for a simple SIP proxy
by AliReza Khoshgoftar Monfared 23 Jun '14

23 Jun '14

Dear Kamailio users, I am trying to set up a simple scenario as follows: UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script: 1) Is there any good example of the above scenario in which the UAC and UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets? 2) In specific, I want the proxies to simply forward the messages to their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration? 3) Let's imagine Proxy_2 in the above example, in the default config, it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3? I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file? Thanks, Alireza

2 4

Is Kamailio PacketCable 2.0 compatible
by Horo, Priyam 23 Jun '14

23 Jun '14

I was looking at installing Kamailio and was wondering it is Packetcable 2.0 compatible? Thanks!

2 1

Disabling rtpengine abrt handler
by Alex Balashov 22 Jun '14

22 Jun '14

Hello, Despite this fix https://github.com/sipwise/rtpengine/commit/cbe1f805363b3d6a117e9e5425d7994… I am continuing to experience periodic crash problems under high loads with rtpengine. This is after upgrading mediaproxy-ng -> rtpengine. Unfortunately, after this upgrade, it is impossible to tell the cause of the crash because the [seemingly] newer rtpengine SIGABRT and/or crash handler is not running properly and is getting in my way: Jun 20 15:47:55 rtphost abrtd: Directory 'ccpp-2014-06-20-15:47:54-1123' creation detected Jun 20 15:47:55 rtphost abrt[2159]: Saved core dump of pid 1123 (/usr/sbin/rtpengine) to /var/spool/abrt/ccpp-2014-06-20-15:47:54-1123 (97013760 bytes) Jun 20 15:47:55 rtphost abrtd: Executable '/usr/sbin/rtpengine' doesn't belong to any package and ProcessUnpackaged is set to 'no' Jun 20 15:47:55 rtphost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2014-06-20-15:47:54-1123' exited with 1 Jun 20 15:47:55 rtphost abrtd: Deleting problem directory '/var/spool/abrt/ccpp-2014-06-20-15:47:54-1123' I am not sure why this script is exiting, but the result is that I have no /var/spool/abrt/ccpp-2014-06-20-15:47:54-1123 and no core dump to work with. This is on a standard minimal CentOS 6.5.x install. Is there any way to either disable this signal handler and just let it dump core normally, or, otherwise, to definitively fix the crash handler? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

3 7

MSILO storing wrong content-type.
by Prasun Bheri 22 Jun '14

22 Jun '14

Hi All, I have Kamailio 4.0.0 configured with MSILO in centos 6. When a Message destined to offline recipient is received by kamailio, MSILO stores the message with wrong content type. In my scenario Incoming message has content type "message/cpim" where as in database its is stored as "text/plain". and obviously when recipient registers back in the stored message is sent with content type as "text/plain". I have attached a wireshark capture where in you can see packet #23 (recived Message) and #88 (dispatched Message). Could you guys let me know how to fix this. Thanks & Regards, Prasun

1 0

UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 21 Jun '14

21 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

1 0

Kamailio clean unconfirmed dialog
by Arun Kumar 21 Jun '14

21 Jun '14

Hi Server : kamailio 4.0 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Dialog TimeOut : 600 (Seconds) Problem : Dialog's are not cleared sometimes , So after sometime server is not responding (out of memory error), After investigating long time , the problem is non confirmed dialog's are not cleared even dialog timeout Thanks Arunkumar

2 4

Re: [SR-Users] SIP Trunk
by Keith 20 Jun '14

20 Jun '14

Hi, So when I use the uac_reg_request_to, for some reason it sends an invite and not a register?? This means I'm not getting a 407 back to the use sac_auth in the failure route. Any ideas? Thanks Keith

1 0

mediaproxy-ng query
by Alex Balashov 20 Jun '14

20 Jun '14

Hello, I'm invoking mediaproxy-ng with rtpproxy_offer/answer("ow") and am getting a scenario where: INVITE is forwarded to signalling gateway xxx.xxx.xxx.xxx, which returns an SDP answer of yyy.yyy.yyy.yyy:51964. rtpproxy is invoked in both directions. However, media forwarded from media endpoint A is sent to xxx.xxx.xxx.xxx:51964 (the correct port, but to the signalling gateway), rather than yyy.yyy.yyy.yyy:51964, which is what's specified in the SDP answer in the 200 OK from xxx.xxx.xxx.xxx. No media is coming back to mediaproxy-ng from yyy.yyy.yyy.yyy:51964. The cause of this is a separate and logically unrelated issue. Nevertheless, I am curious as to why mediaproxy-ng doesn't send media from the near end to yyy.yyy.yyy.yyy:51964--in other words, to the specified SDP endpoint on the far end, rather than to the signalling gateway. I initially imagined that this might be due to the "w" option to rtpproxy_offer(), which the documentation says forces symmetric RTP. However, this behaviour is not elaborated upon or defined. I tried removing the 'w' option but seem to be getting the same result. Is this a default behaviour of COMEDIA-style source port latching? In other words, does mediaproxy-ng need to latch onto incoming RTP from yyy.yyy.yyy.yyy:51964 before it will send media there, as opposed to the signalling gateway? If so, is there any way this behaviour can be disabled, so that mediaproxy-ng simply follows SDP in all cases? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

Kamailio b2bua
by Grant Bagdasarian 20 Jun '14

20 Jun '14

Hello, OpenSIPS has a module for B2BUA. Does Kamailio have a similar one or one that comes close to it? Perhaps by combining modules to make Kamailio act as a B2BUA?

3 4

Initial Install
by Phil Daws 20 Jun '14

20 Jun '14

Hello and good evening all! Am trying to build a Kamailio installation in my lab and have hit the first hurdle; database build. I have a three node MariaDB cluster in place and attempting to get kamdbctl create to set up the tables. What seems apparent is that command requires the database to not already exist. Where can I find the necessary schema to load into MariaDB myself please ? Thank you. Phil

2 2

Setting priority (q) for kamctl ul add
by David Wilson 20 Jun '14

20 Jun '14

Hi All, I'm trying to add a permanent usrloc entry via kamctl ul add. This works, but the created entry has a q value of 1.0 which is higher than I need. Is there a way to either: 1. Specify a q value when using kamctl ul add, or 2. Edit the q value of an existing record by using a kamctl command. Cheers, Dave.

3 2

Starting Kamailio SIP server
by Horo, Priyam 19 Jun '14

19 Jun '14

Hi, I am following the quick-start installation guide at http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x and I am stuck when I start the kamailio, using /etc/init.d/kamailio start it gives me an error of "invalid configuration file". I am using the /usr/local/etc/kamailio/kamailio.cfg file which came along with tar.gz file. I do not want sql and no authentication to begin with. What else needs to be modified or is required in the config file. Thanks!

1 0

Error codes with management commands
by PIERRE Laurent 19 Jun '14

19 Jun '14

Hello, It's unfortunate that error codes are not included with management commands. For example, if you tape " kamctl diaplan rm rule 2 999 " instead of " kamctl diaplan rmrule 2 999 " then all the entries in dialplan table will be deleted. In fat the system perform "kamctl diaplan rm" It's very annoying . -- Laurent PIERRE http://www.linkedin.com/in/lpierre

1 0

Unable To Start MSRP Session
by Megha Patil 19 Jun '14

19 Jun '14

Hi, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error(Reason: SIP ;text="media stream failed to start" ;cause=500). What might be the problem?? Does it mean that kamailio can not handle msrp session? If it can handle, how to enable it? I have attached the captures with this mail. Please help me in this regard.

1 0

Re: [SR-Users] SIP Trunk
by Keith 19 Jun '14

19 Jun '14

Hi, Thanks Daniel, however don't I need to send a register to get the 401 back? If so how do I do this bit? Thanks Keith

2 1

Possible Oversight, but CDRs are not being written
by Joel White 19 Jun '14

19 Jun '14

I made a significant change to the way the system I am building is designed. I removed the Database to a redundant cluster and not on the same server as the kamailio install. I changed all references to the DBURL to reflect the IP of the new cluster. I also changed the SQLOPS statement for CDRs to reflect the new IP modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@10.11.12.5/kamailio") All ACC transactions are being recorded to the new DB and the configuration is running well with the exception of CDRs. What am I missing that would prevent writing of CDRs to the DB? I did check connectivity from the Kamailio server to the DB server as well as access to the cdrs table

2 3

Cisco 79XX Series Phones and TLS integration with kamailio
by Jonathan Hunter 19 Jun '14

19 Jun '14

Hi All, As you guys might remember I have been doing alot of work with these legacy handsets recently, in particular the Cisco IP phones 7945G and 7965G. They work well now with kamailio, using UDP or TCP as the transport protocol. I am now looking to implement SIP over TLS with them, and wondered if anyone had completed the same, as it appears they were designed very much to support SIP over TLS and SRTP but with Cisco call manager and not other SIP devices. I am trying to understand if its possible to integrate with kamailio, as from Cisco documentation it appears a CTL file (client trust list) is required which seems to be generated by cisco software, and the handset needs it before it will try to initiate a SIP connection over TLS. I have implemented TLS /SRTP with Cisco SPA's, Bria etc on kamailio fine, so its more a question around the handset and if anyone has achieved this. Thanks Jon

3 8

Internal server error
by Deepak Bhagwat 19 Jun '14

19 Jun '14

Hi all, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error. What might be the problem?? Does it mean that kamailio can not handle msrp? If yes how to enable it in the server? Packets captured are attached. Please help me in this regard. -- Thanks Deepak Bhagwat

1 0

dialog module -- timeout parameter
by Måns Nilsson 19 Jun '14

19 Jun '14

Hi, I need to allow arbitrarily long calls while still using the dialog module. The module documentation does not mention a maxiumum value (2^32 sec?) nor a setting to disable the timeout functionality. I'd guess that setting it to 0 is a common convention in such cases, but I'd rather get it verified. Anything above 10 years is probably a safe bet, so a 32-bit unsigned int is quite fine. :-) Is there a max value? Is there a way to disable the check? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 I think I'll KILL myself by leaping out of this 14th STORY WINDOW while reading ERICA JONG'S poetry!!

2 1

One way Audio with kamailio -Rtpproxy
by master voip 18 Jun '14

18 Jun '14

Hi can some one help me . I am using kamailio 4.1.3 and having one way audio issue. Scenario is Pjsip 2.0 based UA (private IP)------->Router(publicIP)--------> Kamailio (with RTP proxy)-----------> Third Party_Sip Server ------>PSTN UA is registering fine with sip server but when i make a call they can hear but i can not hear the other party . What i found is in SDP first c=IN is private ip of UA and second c=IN is public ip of UA i dont know if this is the cause When I trace the call on sip server , sip server is sending RTP to UA private ip so it is not reaching kamailio back to forward to UA . I am attaching the kamailio configuration file as well as SIP/SDP trace user agent private ip : 192.168.1.4 User agent public ip : 61.61.61.61 Kamailio IP : 81.81.81.81 Sip Server IP: 71.71.71.71 (* ip addresses are not actual ips) Please reply with detailed instructions to fix this isue config file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # #!define WITH_NAT # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ' '; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ' '; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=3 log_stderror=no #!endif memdbg=1 memlog=1 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=7878 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "nat_traversal.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 7) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication # Below line commented by test-MATHUR as PC-2-PC calls are not getting through #route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { sl_send_reply("100", "Trying"); setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("testM : Unable to relay !!!! :{ \n"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp )\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$s i:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { #xlog("testM : Got BYE from some PSTN/UAC.\n"); setflag(FLT_ACC); # do accounting ... #xlog("testM : Did accounting.\n"); setflag(FLT_ACCFAILED); # ... even if the transa ction fails #xlog("testM : Calling route{NATMANAGE} from WI THINDLG.\n"); #t_newtran(); #t_reply("200", "OK"); #xlog("testM : Called route{NATMANAGE} for BYE froom WITHINDLG.\n"); #exit; } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per R FC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... i gnore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } #if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) #{ # rtpproxy_manage(); # return; #} rtpproxy_manage("z90"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { add_rr_param(";nat=yes"); } } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n" ); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bu g in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_i p) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------------------------------------------------ SIP TRACE REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62147 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 P-hint: outbound SIP/2.0 407 Proxy Authentication Required CSeq: 62147 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141058 Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301389817101016806045824224" Content-Length: 0 REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62148 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301389817101016806045824224", uri="sip:71.71.71.71", response="0ca765f266b6bfc8ea0ef00227272393" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 62148 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141059 Contact: "57778" <sip:test@192.168.1.4:58931;ob>;expires=600 Expires: 600 Content-Length: 0 SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10987 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Content-Length: 0 P-hint: outbound SIP/2.0 401 Unauthorised CSeq: 10987 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141000 Content-Length: 0 WWW-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301390017101016806050024224" Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10988 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301390017101016806050024224", uri="sip:test@71.71.71.71", response="15ee3de757c18e084fca607438017093" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 10988 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141001 Expires: 3600 Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> NOTIFY sip:test@71.71.71.71 SIP/2.0 CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Contact: <sip:71.71.71.71:5060;transport=udp> Subscription-State: active Event: message-summary Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:test@71.71.71.71 SIP/2.0 404 Not here CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060;rport=5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: pj_arubaslim-16/r2 Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 23382 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301391017101016806051024224" Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> ACK sip:919000000002@71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 ACK Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: pj_arubaslim-16/r2 Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301391017101016806051024224", uri="sip:919000000002@71.71.71.71", response="4edf5e2efd1e8e03bdbe4a693fdfee4c" Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.e624c28002d41f48ae76813b26e516f3.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJ8XsuToNj.92GHJLiuK-JZ5T5PpfTbhr Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 ACK Content-Length: 0 BYE sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23384 BYE User-Agent: pj_arubaslim-16/r2 Content-Length: 0 SIP/2.0 200 OK CSeq: 23384 BYE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

1 0

Trigger m_dump() with REGISTER
by Marc M. 18 Jun '14

18 Jun '14

Hi, I have a basic MSILO setup up and running. For the moment every time the client sends a REGISTER, m_dump() is called. I would like to achieve the following: The clients are registering with either 200 or 300 sec expiration time. I would like to dump the messages only when the client is registering for 300 secs. Only a REGISTER with an expiration time of 300 should trigger m_dump(). Can you help me with this? Any help or idea is appreciated. thanks Marc

1 0

Re: [SR-Users] SIP Trunk
by Keith 18 Jun '14

18 Jun '14

Hi, I have a carrier who requires SIP authentication on calls, can anyone point me in the right direction? Normally in Asterisk or Freeswitch you can authenticate a peer, but can't see how to do this in Kamailio. Thanks Keith

2 1

Use of asterisk * in Kamailio routing
by Joel White 18 Jun '14

18 Jun '14

I am having an issue using an asterisk * in the kamailio routing. How would I implement this? Thank you in advance

3 6

JSON RPC Usage in Kamailio
by Jaytirth S Khairatkar 18 Jun '14

18 Jun '14

Hello, Can anyone give me a link or document which explains how to use the JSON and JSON-RPC module in Kamailio SER? I'll greatly appreciate it. Thanks and regards Jaytirth *Disclaimer* ------------------------------------------------------------ Notice: This e-mail and / or attachments (communication) may contain information that is confidential and proprietary to The Tata Power Company Limited, Strategic Engineering Division (Tata Power SED). If you are not the intended recipient of this communication, please do not use or disseminate the information contained therein, notify the sender by e-mail or telephone and delete the communication (permanently) from your system. Any information in this communication that does not relate to the official business of Tata Power SED shall be understood as neither given nor endorsed by it. Please note that e-mails are susceptible to change and Tata Power SED shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt. This communication is not guaranteed to be free from computer viruses and it is recommended that you check for all viruses before download. Print this communication only if absolutely essential. Thank you for your co-operation. ------------------------------------------------------------

1 0

s-cscf - limit of entries for 1 contact
by Ciprus, Daniel 17 Jun '14

17 Jun '14

All, is any of you aware about limitation how many reg entries can be stored in s-cscf for 1 contact ? From what I'm observing here : s-cscf can have up to 12 reg entries for 1 contact and then other entries are silently discarded. Use case: - crash of app happens 12 times - app retries to register - s-cscf sends back 200 OK with 13 contacts - another app registers and tries to send SIP MESSAGE to 1st app Result: message never gets delivered and tcpdump shows 12 messages sent out to p-cscf but not 13th one which is actually live session between client and proxy. thanks Dan ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 2

Install SEMS
by Agiftel 17 Jun '14

17 Jun '14

Hi All, i am trying to figure out how to get SEMS and install it. If i go to iptel website, documentations and last versions give always back NOT FOUND. Is this project still maintaned?/avaible? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Install-SEMS-tp128500.html Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] LCR from URI (CLI)
by Keith 17 Jun '14

17 Jun '14

Hi all, I have an issue where I have two carriers, one only accepts +44 then the CLI and the other only accepts the number with the leading zero dropped. I am using LCR to route to carriers, however I can only switch carriers by manually altering the CLI. Is there a way I can manipulate the CLI based on the destination the call is being sent to? Thanks Keith

2 1

Problems to start kamailio
by Carlos Rangel 17 Jun '14

17 Jun '14

Hello I am having problems to start kamailio 4.1.3 on a fresh centos 6.5 install with Asterisk 11.9 following the tutorial. Everything has gone well except that I cannot start kamailio. Basically I have two problems. The message that I got from the console when I try to start is [root@gt-ev-i08 kamailio-4.1.3]# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed An when I look at the error messages in var/log/messages I see the following errors. Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: missing '(' or ')' ? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: '('')' expected (function call) Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 18: bad command I did not touch that section of the kamailio.cfg file. I am really lost now. Hopefully someone can help. Thanks Carlos

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Unable to make calls within the extensions
by Chandramouli P 17 Jun '14

17 Jun '14

Hi, I am new to Kamailio. I started RTPProxy using "rtpproxy -A 54.85.12.15 -F -l 10.0.0.122 -s udp:localhost:7722" command and see that my sip phones are registered with Kamailio. I am able to see using "kamctl ul show" command. But, I am unable to establish call between my registered extension through Asterisk using Kamailio. I see that calls are hitting my Asterisk server when I call from extension. I am not getting any audio and the other extension is also not at all ringing. I could not able to figure out where I am doing the mistake? I am not sure whether the mistake is in Kamailio or Asterisk. In my Asterisk server, I am seeing the correct configuration using "odbcinst -q -d" and "isql -v MySQL-asterisk chandra test123456" and "odbc show (At CLI>)". Please find the below environment: Operating System: Ubuntu 14.04 Server (64-Bit) Kamailio: 4.0 Asterisk: 11.10 Database: MySQL (UNIX ODBC) Environment: Amazon EC2 Follwed Links: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb http://pastebin.com/VPpfErYn Daniel's Patched RTPProxy Installation (Please note that My Asterisk and Kamailio servers are behind NAT in Amazon EC2) Below is my configuration: *http://pastebin.com/fr8m9gr7 <http://pastebin.com/fr8m9gr7>* *Note:* I inserted the records in to the respective tables with "cmp" as context. When I call from 100 to 500, I am not getting any sound and another extension is not ringing and getting the below messages at Asterisk CLI: *http://pastebin.com/y4tXXrnF <http://pastebin.com/y4tXXrnF>* Any update would be appreciated. Thanks in advance. Regards, Chandra.

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Problem with CANCEL handling
by Mordechay Kaganer 17 Jun '14

17 Jun '14

B.H. Hello, all! I'm trying to set up kamailio as a simple load balancer proxy between PSTN gateway and a cluster of asterisk servers - rather common use case IMHO. I'm completely new to kamailio. I have started with the default kamailio.cfg that comes with Ubuntu package and added the dispatcher module as described in the docs ( http://www.kamailio.org/docs/modules/4.0.x/modules/dispatcher.html). At the end, i got this up and running, but with one exception: if incoming call is not answered by the asterisk end (i.e. it is in ringing state), and then the gateway sends CANCEL request, kamailio is unable to recognize the transaction and is not forwarding the request to asterisk. The result is: the call is left "dangling" in the PBX and is not properly hang up. When the gateway sends CANCEL to kamailio, i get the following log messages: 4(25442) DEBUG: tm [t_lookup.c:769]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: searching on hash entry 31287 4(25442) DEBUG: tm [t_lookup.c:492]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(25442) DEBUG: tm [t_lookup.c:859]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: no CANCEL matching found! 4(25442) DEBUG: tm [t_lookup.c:861]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT completed 4(25442) DEBUG: tm [tm.c:1039]: w_t_lookup_cancel(): lookup_original: t_lookupOriginalT returned: (nil) 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=485 a=25 n=xdbg 4(25442) DEBUG: <script>: is_method(CANCEL): t_check_trans()==false 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=487 a=2 n=exit The relevant script snippet is standard from the distro package: # CANCEL processing if (is_method("CANCEL")) { xdbg("is_method(CANCEL): entry"); if (t_check_trans()) { xdbg("is_method(CANCEL): t_check_trans()==true"); route(RELAY); } else { xdbg("is_method(CANCEL): t_check_trans()==false"); } exit; } How can i fix this? Thanks! -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!

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Re: [SR-Users] sip aliases
by Anna Schuster 17 Jun '14

17 Jun '14

Hi Quick question: If I create different sip aliases under my account, is any way I can see under Missed Called section on serweb.iptel.org or any elsewhere on which alias the call came on? Thank you, Anna On Sat, Jun 7, 2014 at 3:11 PM, <registrar(a)iptel.org> wrote: > > Thank you for registering with iptel.org. > > We are reserving the following SIP address for you: sip:mielul@iptel.org > > To finalize your registration please check the following URL within 24 > hours: > http://serweb.iptel.org/user/reg/confirmation.php?nr=0eb4729fad0f99434d205f… > (If you confirm later you will have to re-register.) > > Here are the details of your account: > --------------------------------------------------------------------- > Username: mielul(a)iptel.org > Password: q1q1q1q1 > Email: annasc74(a)gmail.com > Full Name: Anna M > Sip address: sip:mielul@iptel.org > (You can always retrieve these via the "Forgot Password" link on the > signup page) > > > > -- This message is automatically generated by SerWeb. >

2 1

Re: [SR-Users] exec_avp problem
by Ansuman Manna - Asergis Telecom 17 Jun '14

17 Jun '14

Hi, I am using the exec_avp command as followed: exec_avp("/usr/bin/mwi/test.sh $fU"); $fU taking command line arguments. Here is the shell script content: #!/bin/bash -e echo $1 logger done! $1 logger done mkdir /usr/bin/mwi/$1 || { logger 'touch failed' $? $errorMessage ; exit 1; } logger is working fine but.mkdir and other system commands are not executing. I cant figureout my misses Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

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How to use unixodbc/freetds for kamailio backend
by Tim Chubb 17 Jun '14

17 Jun '14

Hi Im just getting started with kamailio with a view of evaluating it for production use. Is it possible to do the following: 1) I would like to use ODBC/FreeTDS to connect to an existing ms sql server cluster and use it for all DB related activity relating to kamailio, is this possible as we would be running kamailio in a HA configuration in production and I really don't want to have to support another DB cluster? 2) Looking at the kamdbctl scripts it seems that currently every popular db but sql server is supported which is a bit of a shame, however am I right in saying that if I was to port the sql files to SQL server, and implemented a script similar to kamdbctl.pgsql to wrap around the freetds TSQL executable that would be all that I would need to do add support for SQL server? 3) Is there any particular reason why there isn't an ODBC option for the kamdbctl script? Cheers.

2 1

Kamailio and Freepbx Integration
by Carlos Rangel 16 Jun '14

16 Jun '14

Hello List I am trying to intgrate Kamailio latest release 4.1.4 and Freepbx 5.2.11 Asterisk 11.0.9 in the same box on an internal network following Daniel´s tutorial (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ) . I am very new with Kamailio and I tried like 15 times, seriously. The difference is that I am not using the sipusers table nor sipregs (I may later use sipregs) nor the voicemail tables (Freepbx can manage voicemail). I believe I was able to modify the kamailio.cfg file (DBURL and DBASTURL) in order to fetch the user information from the Freepbx tables. There is a table called devices that contents all the information needed. I also managed to not fill the secret parameter (password) on the extension configuration so Asterisk does not ask for authentication twice as recommended in the tutorial. I put the password in other table and then tell Kamailio to fetch it from the corresponding table. I can control if the users are registered in the system as long as the user is created in Freepbx with the right password, again password is not on the secret column. The weird thing is that my users can register in Asterisk even if they are not created in Kamailio (kamctl add). I was expecting them not to register if they are not created on the database. I have configured Freepbx to bind to a specific internal IP and port 5080 and configures kamailio as in the post #!ifdef WITH_ASTERISK asterisk.bindip = "xxx.xxx.xxx.xxx" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "xxx.xxx.xxx.xxx" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif Also I have added to kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_ANTIFLOOD I dont think this is the normal behavior, does anybody have an insight of what could be happening?? MY goal is to add the #!define WITH_NAT functionality (I havent done it yet)and then have remote trusted users connected to the PBX. Of course the concerns is to keep hackers away from my system. Thanks in advance for your help and for this great piece of software!!!! Regards and greetings from Mexico Carlos

1 0

stampeeding server.
by Måns Nilsson 16 Jun '14

16 Jun '14

Hi, I've got a 3.3.2 Kamailio that from time to time gets very upset and starts rushing these log entries to syslog: Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore The rate is about 10000 lines per second. This fills the log partition, which locks other things up and the users are unhappy. The server recovers nicely if restarted. I have scarce logs because I have to remove logs to get service back. Any suggestions? More data on request. -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 HELLO, everybody, I'm a HUMAN!!

2 3

Kamailio help
by Nouri El Sebai (Student) 16 Jun '14

16 Jun '14

Dear Sir/Madam, I am trying to connect (in Lebanon) as a subscribed user on Kamailio that has been installed on a remote server in the US. I am being faced with the error: Forbidden (403) Reason: Not relaying Kindly advise what I should do. Best regards, Nouri El Sebai

2 1

rtpproxy_manage and ip addresses
by tonchek＠mac.com 13 Jun '14

13 Jun '14

HI to ALL! As I understand the logic of rpproxy_manage work, it uses packet source IP and port number from m= field of SDP as socket for rtpproxy process. So, we can see it in SIP and rtpproxy debug messages: IP 77.94.133.231.5060 > 77.94.133.232.5060: UDP, length 1170 [...headers stripped...] v=0 o=root 1523540081 1523540081 IN IP4 188.40.66.75 s=Asterisk PBX 1.6.1.20 c=IN IP4 188.40.66.75 t=0 0 m=audio 15260 RTP/AVP 8 0 101 [...sdp stripped...] Jun 13 21:38:09 puma rtpproxy[4728]: INFO:handle_command: new session on a port 57680 created, tag FppNrF9K7S8gF;1 Jun 13 21:38:09 puma rtpproxy[4728]: INFO:handle_command: pre-filling caller's address with 77.94.133.231:15260 where 172.22.11.16 is SIP packet source IP address and 17646 is the port number from SDP. Is it possible to change this behavior and use IP address from incoming SDP as well as port number? Thank you Anton

1 1

inbound kamailio dispatcher faulty routing of ack after 200 OK
by Gijs Kwakkel 13 Jun '14

13 Jun '14

client1 <> provider <> kamailio <> asterisk <> client2 when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated. after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk. I added the config, syslog and wireshark output (converted)

2 6

Re: [SR-Users] Is t_on_failure with uac_req_send possible?
by Ian Lin 13 Jun '14

13 Jun '14

Hi Klaus, I've got the same problem on kamailio-4.1.0 and I couldn't see the sip messages reach to onreply route while using uac_req_send. Could you please give me a hint about this? Thanks! Cheers, Ian -- ------------------------ Ian Lin 林于巽 ------------------------

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sr-users June 2014