- sr-users - kamailio.org

LCR
by Douglas Ugalde 11 Jan '13

11 Jan '13

Hi, Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I do to fix this error: ERROR: lcr [lcr_mod.c:1840]: error while parsing R-URI This is my LCR configuration block: route[LCR] { if(!load_gws(1)){ sl_send_reply("500", "Internal server error, unable to load gateways"); xlog("L_NOTICE","Internal server error, unable to load gateways"); break; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); break; } } I not sure if this configuration thats ok, please somebody help me. Note: Params and modules are already loaded, sorry for my english. Best regards.

2 1

Problem with MSILO module forwarding messages
by Owen Lynch 10 Jan '13

10 Jan '13

Hi, I have the kamailio 3.3.2 msilo module basically working but I cannot configure it to forward stored messages to the registered UA. Instead it forwards to itself (via DNS lookup on the domain name). This is causing me grief because I am getting recursive message handling. The module overview describes this behaviour: "Every time when a user registers with Kamailio, the module is looking in database for offline messages intended for that user. All of them will be sent to contact address provided in REGISTER request." The REGISTER request contact header looks OK: Contact: <sip:hxqecvtn@192.168.2.236:57532 ;transport=tls>;expires=3600;received="sip:192.168.2.236:53628 ;transport=TLS";+sip.instance="<urn:uuid:1ee63413-3569-46dc-a9ee-34c6e4e1403c>" However the 'outbound_proxy' parameter seems to contradict this: "The SIP address used as next hop when sending the message. Very useful when using Kamailio with a domain name not in DNS, or when using a separate Kamailio instance for msilo processing. If not set, the message will be sent to the address in destination URI." So, when 'outbound_proxy' is not set, the message is sent to the address in the destination URI, rather than the UA contact address? Can the module be configured to forward stored messages directly to the UA contact address? Thanks, Owen Lynch

2 1

Socket error
by Douglas Ugalde 10 Jan '13

10 Jan '13

Hi, Please I need help with this error: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) I'm using Kamailio 3.3.3 I call 2 times before got this error, how can I do to resolved it, I really really appreciated any help. I attached my Kamailio.cfg. Really sorry for my English. Best Regards.

2 1

NAT Problem! Not relay ACK,CANCEL and BYE to other UA
by Nguyen Anh Tuan 10 Jan '13

10 Jan '13

Hi All, I have 2 interfaces - External A (behind NAT) and Internal B (LAN) in Kamailio Load Balance, and enable Multi-Homing function "mhomed=1". Load Balance dispatches to Kamailio SIP Servers via the Internal B (LAN) (each Kamailio SIP server has both proxy and registrar in one box). For RTP relay, I use RTPproxy in other server, has 2 interfaces A&B like Load Balance. If using 2 interfaces in Load Balance, Registration and calling NORMAL with 2 users behind NAT, but the call will be interrupted after 30 seconds due to the ACK message (after Ringing, 200 OK) STOP in Load Balance in return and not relayed to other UA . CANCEL and BYE have same error like ACK. I tried using only one interface for load balancing (External = Internal IP), and SIP Server binds to Load Balance in the same network, EVERYTHING IS OK. ACK, CANCEL and BYE are not missing behind NAT. For NAT Traversal I use PATH Module in Load Balance, and add "Received" like this: ... modparam("path", "use_received", 1) .. .. record_route(); add_path_received(); force_rport(); t_relay; .. .. And in SIP Server enable Path Parameter for Registrar: modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) Enable "#!define WITH_NAT" in default kamailio.cfg for RTP Controlling, but I removed all fix_nated_register and fix_nated_contact in SIP Server, because using PATH module.I try enable both fix_nat_register and fix_nated_contact, or only fix_nat_register but NOT SOLVE Problem "missing ACK,CANCEL, and BYE in Load Balance" behind NAT. Best Regard, Tuan.

2 1

Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
by Richard Brady 10 Jan '13

10 Jan '13

Hi Peter Great work on this! We'd like to help you test. The test I have in mind, which we could create using SIPp, would be to register multiple contacts with the same instance-id (i.e. "sip.instance" param) but different reg-id params. Then send an INVITE to that AoR and make sure the forking is only per instance-id and not per reg-id. This could be repeated in multiple permutations of instance-ids and reg-ids. This would be a test of save() and lookup() more than anything else. Is that what you had in mind? Richard On 3 January 2013 14:13, Peter Dunkley <peter.dunkley(a)crocodile-rcs.com>wrote: > I hope to get the outbound edge proxy and flow timer stuff into master by > Monday, but it could really do with some additional testing especially in > conjunction with the reg-id stuff in registrar/usrloc (which I have no idea > how to use). > > Peter > >

5 14

LCR with Siremis
by Douglas Ugalde 09 Jan '13

09 Jan '13

Hi, I was configuring the modparams for LCR and Dialplan, I taken from kamailio web site, but I think the params are outdated, because we need to erase some lines to match with the tables in the DB, Im using Kamailio 3.3.3 and Siremis 3.3 Please can you give 1 example about how to make outbounds calls with siremis 3.3 (send calls to gateway)? I already have the lcr and dialplan module loaded with params. Where can I get update documentation about Siremis 3.3 and Kamailio 3.3.3. Thanks for your help.

1 0

Freezing development for v4.0.0
by Daniel-Constantin Mierla 09 Jan '13

09 Jan '13

Hello, a (26-hours before) reminder that development for next major release v4.0.0 will be frozen tomorrow, Thursday, January 10, 2013, at 23.59 GMT. If any developer has new features to push in the git master branch, please do it before. After that, for a while, the master should get only bug fixes and improvements to documentation and addition tools (e.g., bringing kamctl up to date). New features have to be kept in personal branches until the master branch is open again. The focus should be on testing from the day after tomorrow on, I hope the users community will help us here as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Having "modparam" options in the database
by Glenn O. Larsen 09 Jan '13

09 Jan '13

On Tue, Jan 8, 2013 at 9:21 AM, Olle E. Johansson <oej at edvina.net> wrote: >>>> Is there a way to read modparam config from a database? > In most cases, there should be no need to change modparams at > runtime. > > Can you tell us more about which parameter you want to change during > runtime that doesn't have a mI/rpc reload command? Hi Olle! The module we have been using, and wanted database configuration for was the pike module. But and after spending some more time looking at alternative modules, I see that we can use other modules to help us with traffic control/blocking, like the pipelimit and the ratelimit (both these modules have some sort of database integration, need to look more into this). I was so focused on the pike module, that I didn't see these other modules.. :) -- mvh Glenn O Larsen

3 3

Core fm_realloc function
by Bruno Bresciani 08 Jan '13

08 Jan '13

Hi all, I observed two at 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536. I don't know exactly what may be caused this core because the debug of kamailio is disabled, but looking at backtrace (gdb) and source code I could notice a relationship with TLS connection establishment (handshake). Other important fact is that core was generated by the kamalio itself with signal 6 (abort) because it can't find the pointer to memory realloc. I'd like to know if someone already observed this issue and if my analyse above is correct? Bellow is backtrace full with gdb: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00f30402 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 0x00f30402 in __kernel_vsyscall () No symbol table info available. #1 0x009fec10 in raise () from /lib/libc.so.6 No symbol table info available. #2 0x00a00521 in abort () from /lib/libc.so.6 No symbol table info available. #3 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536 f = (struct fm_frag *) 0xb61f0bb8 pf = <value optimized out> orig_size = 18 L=ÃbÂ¬Ã2n = (struct fm_frag *) 0xb61f0bd2 ptr = <value optimized out> hash = 23548 #4 0x0051a87f in ser_realloc (ptr=0xb61f0bc0, size=32) at ../../mem/shm_mem.h:266 No locals. #5 0x00460295 in OPENSSL_cleanse () from /lib/libcrypto.so.6 No symbol table info available. #6 0x004609f7 in CRYPTO_realloc () from /lib/libcrypto.so.6 No symbol table info available. #7 0x003efb22 in sk_insert () from /lib/libcrypto.so.6 No symbol table info available. #8 0x003efbaa in sk_push () from /lib/libcrypto.so.6 No symbol table info available. #9 0x007c2ac3 in ssl_bytes_to_cipher_list () from /lib/libssl.so.6 No symbol table info available. #10 0x007aa9f3 in ssl3_get_client_hello () from /lib/libssl.so.6 No symbol table info available. #11 0x007ab815 in ssl3_accept () from /lib/libssl.so.6 No symbol table info available. #12 0x007c1c3a in SSL_accept () from /lib/libssl.so.6 No symbol table info available. #13 0x005278e3 in tls_accept (c=0xb621af88, error=0xbf91dfc8) at tls_server.c:346 ret = <value optimized out> ssl = (SSL *) 0xb61c1878 cert = <value optimized out> tls_c = (struct tls_extra_data *) 0xb6208e58 tls_log = <value optimized out> #14 0x0052da8e in tls_read_f (c=0xb621af88, flags=0xbf91e200) at tls_server.c:1028 r = (struct tcp_req *) 0xb621affc bytes_free = 4095 bytes_read = 70 read_size = 4095 ssl_error = 0 ssl_read = 0 ssl = (SSL *) 0xb61c1878 rd_buf = "\026\003\001\000A\001\000\000=\003\001OÃ¿\vÂ¡ywP\fkÃDÃ 0/\000\a\000\005\001\000ÃÃÃÃ¯000\000\026\0009\0008\0005\000\026\000\023\000\n\0003\0002\00 202i<Ã¤<Ã¥Ã¾\031Ã¬234Â \237Ãª030Y){lÑ¶"...06Ã'Ãr9Ã¬Ã«Â¸{[Â©\t\205\037\036Ã¿HPH5Âµ\t(Ã©32Ã027Ã¦Ã²036Â¾\"\rÔ¤YÃ215uÂº\027)Ã¶000Ã±215\vÃÃ Ã©Ea\023xÃ©J\206T\204CÃ wr_buf = "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"... rd = {buf = 0xbf90dfa4 "\026\003\001", pos = 70, used = 70, size = 65536} wr = { buf = 0xbf8fdfa4 "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"..., pos = 0, used = 0, size = 65536} tls_c = (struct tls_extra_data *) 0xb6208e58 enc_rd_buf = (struct tls_rd_buf *) 0x0 n = 0 flush_flags = <value optimized out> err_src = 0x541a83 "TLS read:" x = <value optimized out> #15 0x08158ade in tcp_read_headers (c=0xb621af88, read_flags=0xbf91e200) at tcp_read.c:406 bytes = -1 remaining = <value optimized out> p = <value optimized out> r = (struct tcp_req *) 0xb621affc #16 0x08158fd4 in tcp_read_req (con=0xb621af88, bytes_read=0xbf91e204, read_flags=0xbf91e200) at tcp_read.c:871 bytes = <value optimized out> total_bytes = 0 resp = <value optimized out> size = <value optimized out> req = (struct tcp_req *) 0xb621affc dst = {send_sock = 0x1, to = {s = {sa_family = 57540, sa_data = "\221Â¿\020\000\000\000\000\000\000\000\020\000\000"}, sin = {sin_family = 57540, sin_port = 49041, sin_addr = {s_addr = 16}, sin_zero = "\000\000\000\000\020\000\000"}, sin6 = {sin6_family = 57540, sin6_port = 49041, sin6_flowinfo = 16, sin6_addr = {in6_u = {u6_addr8 = "\000\000\000\000\020\000\000\000\001\000\000\000\001\000\000", u6_addr16 = {0, 0, 16, 0, 1, 0, 1, 0}, u6_addr32 = {0, 16, 1, 1}}}, sin6_scope_id = 12}}, id = -1080958468, proto = 4 '\004', send_flags = {f = 0 '\0', blst_imask = 0 '\0'}} c = 10 '\n' #17 0x08159a0f in handle_io (fm=0x83296b0, events=1, idx=-1) at tcp_read.c:1150 ret = 4 n = 4 read_flags = 1 con = (struct tcp_connection *) 0xb621af88 s = 12 resp = <value optimized out> t = <value optimized out> #18 0x0815d653 in tcp_receive_loop (unix_sock=35) at io_wait.h:1091 No locals. ---Type <return> to continue, or q <return> to quit--- #19 0x0812885b in tcp_init_children () at tcp_main.c:4819 r = 3 reader_fd_1 = 35 pid = <value optimized out> si = <value optimized out> #20 0x080ad412 in main_loop () at main.c:1632 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=192.168.166.31:5060 \000\221Â¿\002\000\000\000/\200\220\000\001\000\000\000\000\000H\000Â¨Ã¤21Â¿\001\000\000\000Â°Â¶Ã¨026Â¶\000\000\000\000\000ÃÂµ\030s2\b\002\000\000\000Ã¢f\b\000ÃÂµ\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000Ã¿Ã¿Ã¿Ã¿Â¸Ã¤21Â¿" #21 0x080afbe7 in main (argc=3, argv=dwarf2_read_address: Corrupted DWARF expression. ) at main.c:2398 cfg_stream = (FILE *) 0x8a2d008 c = <value optimized out> r = 0 tmp = 0xbf91e674 "\212Ã¾\221Â¿\233Ã¾\221Â¿Â°Ã¾\221Â¿Â»Ã¾\221Â¿Ã\221Â¿Ã¹1Â¿\020Ã¿\221Â¿DÃ¿\221Â¿LÃ¿\221Â¿WÃ¿\221Â¿]Ã¿\221Â¿oÃ¿\221Â¿{Ã¿\221Â¿\202Ã¿\221 tmp_len = 10492229 port = 136192361 proto = -1080957480 ret = <value optimized out> seed = 21782655 rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> Best Regards

4 24

ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES)
by Shreyansh Purwar 08 Jan '13

08 Jan '13

Hello Everyone, While creating an account on Kamailio Sip server (I am running the command "kamctl add test test1"), it is asking for password of MySQL Server and after entering the password, I am getting following errors :- - ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) - ERROR: introducing the new user 'test' to the database failed Please help me in resolving this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

raw_query is crashing....
by Krishna Kurapati 08 Jan '13

08 Jan '13

Hi, Since there is no way to limit the result set to 1, I tried to use raw_query inside msilo module. Everytime it executes, it is crashing. The same query if I run in mysql shell, it is returning successfully. Here is the code That I added: query_len = snprintf(query, 2048, "SELECT * FROM %.*s WHERE %.*s='%.*s' AND %.*s='%.*s' AND %.*s!=%i AND %.*s<%i LIMIT 1;", ms_db_table.len, ms_db_table.s, sc_uri_touser.len, sc_uri_touser.s, puri.user.len, puri.user.s, sc_uri_tohost.len, sc_uri_tohost.s, puri.host.len, puri.host.s, sc_status.len, sc_status.s, 200, sc_status.len, sc_status.s, MSG_NOTIFY_OFFSET); if (query_len < 0) { LM_ERR("error in sql snprintf"); goto error; } LM_DBG("m_dump_msg: Query=%s\n", query); query_str.s = query; query_str.len = query_len; if (msilo_dbf.raw_query(db_con, &query_str, &db_res) < 0) { LM_ERR("Failed to query database. Query = %s\n", query); goto error; } The Debug Log at Level 4 shows following: 11(7520) DEBUG: msilo [msilo.c:1120]: m_dump_msg: Query=SELECT * FROM silo WHERE tousername='968257197' AND todomain='50.62.1.7' AND status!=200 AND status<1000 LIMIT 1; 11(7520) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x7fdd4b605540 11(7520) DEBUG: db_mysql [km_res.c:68]: 16 columns returned from the query 11(7520) DEBUG: <core> [db_res.c:155]: allocate 128 bytes for result names at 0x7fdd4b5d8dc0 11(7520) DEBUG: <core> [db_res.c:165]: allocate 64 bytes for result types at 0x7fdd4b5d8e60 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x7fdd4b5d8d90 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d90)[0]=[id] 11(7520) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x7fdd4b5d8d60 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d60)[1]=[src_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x7fdd4b5d8d30 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d30)[2]=[dst_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type <skip> 11(7520) DEBUG: <core> [db_val.c:117]: converting STRING [ac3aeddd9b6f45e6822c824aa2c133b2] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [0] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [1] 26(7552) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 22 26(7552) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 11, pid 7520 (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 22, -1, 0x0) fd_no=44 called 26(7552) DEBUG: <core> [tcp_main.c:3353]: DBG: handle_tcp_child: dead tcp child 1 (pid 7520, no 11) (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 25, -1, 0x0) fd_no=43 called 0(7509) ALERT: <core> [main.c:787]: child process 7520 exited by a signal 11 0(7509) ALERT: <core> [main.c:790]: core was not generated 0(7509) INFO: <core> [main.c:802]: INFO: terminating due to SIGCHLD As you can see something happening to Database connection. Not sure why I am unable to use raw_query. Normal API query from msilo works just fine. What am I doing wrong to make it crash? Thanks Krish Kura

2 1

Having "modparam" options in the database
by Glenn O Larsen 08 Jan '13

08 Jan '13

Hi guys, I have tried to Google this for some time now, but can't seem to find an answer. Is there a way to read modparam config from a database? -- Best regards, Glenn Larsen

3 3

LCR weight
by Ricardo Martinez 08 Jan '13

08 Jan '13

Hello. I have a doubt about the WEIGHT value in the LCR module. In the past we used the Kamailio version 1.5.2 and in that version LCR weight was calculated by a little script called lcr_weight_script.php. There was not “percent” values like 50% and 50% for two gateways despite that the probabilities were numbers calculated by the script. My question is : For Kamailio version 3.2.4 do I have to use the same script to calculate the weight?? I was almost sure that the weight column was a probability value. For example if I have 3 gateways, with 50% 25% and 25% The weight column should be : 50, 25 and 25 ? or 45, 32 and 32 as the script indicates??? [root@vlcr utils]# ./lcr_weight_test.sh 45 32 32 weight 45 probability 0.5235 weight 32 probability 0.2352 weight 32 probability 0.2413 Please enlighten me here. Thanks Ricardo.-

2 1

TCP Path received quotes
by Spencer Thomason 07 Jan '13

07 Jan '13

Hello, I'm forwarding registrations to Freeswitch and adding a Path header using add_path_received() for endpoints behind NAT. When TCP is used, this results in a header like this: Path: <sip:a.a.a.a:5070;lr;received="sip:b.b.b.b:35430;transport=tcp"> Freeswitch doesn't like the transport protocol inside the quotations and refuses to route a call to the registered user. Constructing the header manually without the quotes such as: append_hf("Path: <sip:$Ri:$Rp;lr;received=sip:$si:$sp;transport=$pr>\r\n"); solves the problem. Is there a reason for the quotes? Best Regards, Spencer

4 7

Multi tenant Kamailio
by Rumen Mihailov 07 Jan '13

07 Jan '13

Hello Kamailio, VoIP and asterisk gurus, I have the following really complicated scenario to solve. I have CompanyA which has extensions 100-120 I have CompanyB which has extensions 100-115 I have CompanyC which has extensions 100-130 Now I want to have all those extensions on a single kamailio box, but I want to achieve the following. CompanyA calls 110 and this rings CompanyA 110 not CompanyB or CompanyC. What I want to achive is multitenancy. I know how to make this in asterisk but I want to take advantage of Kamailio's load balancing and HA features. Sammy from the mailing lists advised that I use multidomain and actually I saw that this module is indeed intended to do that, however I have no clue how to use it. I mean do I have to setup a domain for all the companies ? For instance companya.itradebg.com, companyb.itradebg.com and then advise the customers to register to their respective domain or what ? This concept is really a bit tough for me despite all the help from Sammy for which I am really greatful. I would greatly appreciate any tutorials or links to some useful documentation apart from the module documentation. Thank you in advance guys. Best regards, Rumen http://itradebg.com

2 2

NAT'ed RLS setup
by Dmytro Bogovych 07 Jan '13

07 Jan '13

Greetings. I have kamailio 3.3.3 setup on server behind the nat. Nat forwards ports for tls/tcp/udp. This setup worked without rls ok. I added necessary modules - rls, pua, xcap etc. Updated a config file and db. Client application uploads xcap documents ok; after app sends subscribe request to list address. Rls/pua modules generate subscriptions to entries in list. But dns resolver finds the public ip as target for subscription. The result is: sip packets cannot reach the server - they are sent to public ip of nat machine. I see it in ngrep output. Nat does not route them back. I add the domain name and _sip._udp.domain with internal ip address to /etc/hosts. But kamailio does not use /etc/hosts when dns caching is active. I do not want to loose dns caching. I made an attempt to run without caching but server was not stable. It crashed couple of times - so i stopped these attempts. Is there way to put eternal domain -> ip bindings into dns cache? I want to put bingings to internal ip - hope it will help. Maybe there is another way to handle this problem?

1 0

Timeout for t_uac_dlg per request
by David 07 Jan '13

07 Jan '13

Hello, I am running Kamailio 3.2.x, I am using udp mi to send a t_uac_dlg command, :t_uac_dlg: NOTIFY sip:phone1@proxy.lan . . "From:sip:daemon@proxy.lan To:<sip:phone1@proxy.lan> Contact:sip:daemon@proxy.lan" # I would like to know if I can change the timeout for this request. Running a packet sniffer on the proxy, I see the packet is retransmitted after 500ms, then 1000ms, then 2000ms, etc... This is the proper behaviour for SIP, so we can see my tm module is working well. My problem is that I want a response within 3 seconds ( I know this doesn't respect the normal SIP timer values ). I do not want to change the timers in the tm module for the entire proxy, so my question is if there is a way to set the timeout for the dialog created by the t_uac_dlg command, and can I set it on a per dialog basis ? David

1 0

TLS with self-singned certificates
by Iurii Andamasov 07 Jan '13

07 Jan '13

Hello, trying setup kamailio with TLS, have 2 peers, tls.cfg: http://pastebin.com/DvdDzx0v i'm getting Jan 4 10:38:43 fs-tls /usr/sbin/kamailio[3798]: ERROR: tls [tls_server.c:1190]: TLS read:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure Can someone point me to solution? Thanks

3 4

tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory
by Shreyansh Purwar 07 Jan '13

07 Jan '13

Hello Everyone, While building kamailio server 3.0.0 with TLS(I have removed tls from exclude module variable) I am getting following error :- - tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory Please help me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Malicious traffic comming from 37.75.210.90
by Nick Khamis 06 Jan '13

06 Jan '13

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.

1 0

dialog ka_timer and ka_interval problem
by Rinor Hoxha 06 Jan '13

06 Jan '13

Hi, I had an issue with dialog module of kamailio v3.3.3. Is the same described on http://lists.sip-router.org/pipermail/sr-users/2012-November/075735.html. I think the issue is related to dgl_hash.c on line:192 *dlg_ka_list_head = NULL; should be *dlg_ka_list_tail = NULL; and line:216 *dlg_ka_list_tail = dka; should be *dlg_ka_list_head = dka;. I will test it as soon as I can, but meanwhile I'm attaching the patch that should fix it so you can test it and report back the results if you want to. Br, Rinor

2 1

Kamailio DB version error
by Douglas Ugalde 06 Jan '13

06 Jan '13

Hi, I want to load the "dialplan.so" with parameters but when I reload kamailio, I got this error: /usr/local/sbin/kamailio[1348]: ERROR: <core> [db.c:439]: invalid version 0 for table my_table found, expected 1 (check table structure and table "version") Im using kamailio 3.3.3 Please what can I do to resolve it? Regards.

2 1

Next major release proposed as v4.0.0
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, using v4.0.0 for the next major release was proposed and debated at previous development IRC meeting. First was about the addition of a new transport layer (websockets) along with other important new features. During the debate was touched the completion of kamailio-ser integration (no more duplicated modules) as another milestone for increasing the first digit (which was finished just before the end of last year). Now we also have the IMS modules, pushed in the GIT master branch few days ago. It seems that there are lot of good reasons for numbering the next major version 4.0.0. Other comments, opinions? Have a great and happy year ahead, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 4

Multiple rtimer processes + mqueue
by Alex Balashov 04 Jan '13

04 Jan '13

Will multiple rtimer processes consuming the same mqueue, i.e. via a blocking mechanism while(mq_fetch(...)) ... hurt anything? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Freezing for next major release
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, based on last irc devel meeting and feedback afterwards, most of the people seem to be fine releasing next major release a bit earlier, in order to: - make websockets support and the other new features available in a stable version - allow proper development time frame for other planned new features that wouldn't have been ready if the release would be 1-2 months later So, considering the winter holidays, I propose to freeze development on January 7, 2013, with full release about one month later. It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 5

"maximum number of branches exceeded" after kamalilio restart
by Andrew Pogrebennyk 04 Jan '13

04 Jan '13

Hello, I'm getting some errors - if I restart kamailio on a live system, the first call after restart fails with "max nr of branches exceeded". However, we are not doing append_branch() or anything fancy for that call at all: Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: <core> [dset.c:306]: max nr of branches exceeded Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: registrar [lookup.c:298]: failed to append a branch Also it appears on the other system with a different config: Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:656]: ERROR: add_uac: maximum number of branches exceeded Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed The second call however goes through and this error does not appear until the next kamailio restart. What could be the problem here? P.S. The kamailio version is 3.3.2. Andrew

1 0

km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory
by Shreyansh Purwar 04 Jan '13

04 Jan '13

Hello Everyone, I need a help, I was installing kamailio SIP server according to the instruction provided at wiki page - http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git but while compiling it(make all) i am getting following error - km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory Can anybody tell me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Re: [SR-Users] [sr-dev] Freezing for next major release
by Peter Dunkley 03 Jan '13

03 Jan '13

I hope to get the outbound edge proxy and flow timer stuff into master by Monday, but it could really do with some additional testing especially in conjunction with the reg-id stuff in registrar/usrloc (which I have no idea how to use). Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >Hello, > >a reminder that the freezing date of the development for the next major >release is next Monday, January 7. > >Considering it was vacation time so far, we can prolong development for >few days if it is needed - just reply here. > >Besides few small additions, I want to add internal hash table to track >MSRP connections, hopefully I should be able to do it in time. If not, I >will ask for few more days as well. > >Cheers, >Daniel > >On 12/5/12 5:05 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> based on last irc devel meeting and feedback afterwards, most of the >> people seem to be fine releasing next major release a bit earlier, in >> order to: >> - make websockets support and the other new features available in a >> stable version >> - allow proper development time frame for other planned new features >> that wouldn't have been ready if the release would be 1-2 months later >> >> So, considering the winter holidays, I propose to freeze development >> on January 7, 2013, with full release about one month later. >> >> It seems that many people are already using devel version to play with >> websockets, meaning that we are staying very well with testing. >> >> Other opinions? >> >> Cheers, >> Daniel >> > >-- >Daniel-Constantin Mierla - http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > >_______________________________________________ >sr-dev mailing list >sr-dev(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

1 0

recreating kamilio_fifo without restart
by Uri Shacked 03 Jan '13

03 Jan '13

Hi, I entered the command "kamailio -v" instead of "kamailio -V". It looks like it starts kamailio again even that it was running.... the problem is that the kamctl_fifo and kamctl_ctl file are gone now. how do i create the files again without restarting kamailio? Thanks, Uri

3 4

All The Best And A Happy 2013!
by Daniel-Constantin Mierla 31 Dec '12

31 Dec '12

This was it for 2012, one of the greatest years in the history of the project! One major release was done in the summer (version 3.3.x), another one is few days before testing phase, most probably to be out by mid of February. The next major release includes several great features and edevelopments: - support for websockets - IMS extensions in the main GIT branch - no more duplicated modules, integration of Kamailio and SER ended There are big plans for 2013. First is the next major release, together with participation at FOSDEM conference in Berlin. A bunch of other events will open the path to our first dedicated conference for Kamailio project - Kamailio World. We hope to meet many of you at Kamailio World conference, in Berlin, Germany, during April 16-17. Last organizing bits were sorted out and registration will open in the first days of 2013 -- you can see more details at: - http://conference.kamailio.com Looking forward to a great 2013 for Kamailio and everyone involved in the project, from developers to community members! All the best! Happy New Year! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio TLS setup and certificate
by Burak ÖZTÜRK 31 Dec '12

31 Dec '12

Hi, I m a new kamailio user. Kamailio version is 3.3. I want to enable kamailio's tls support. I fallowed instructions from this link : http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates And I also enabled tls module in the kamailio.cfg file. (definig WITH_TLS and adding listen=tls:myIP:5061 command) But after trying to start kamailio i got "failed to start" error. When I checked the logs there is nothing about tls or ssl. Actually I got an error while fallowing the link above. "openssl s_client -connect localhost:5061 -tls1" command returns connection failed 111 error. I thought that this is the reason failed to start. But where or what am I doing wrong? I searched the web and hear about SRV records. But there is no explanations about this, on your guide/wiki. I also read this link too but cant get my answer : http://sip-router.org/docbook/sip-router/branch/master/modules/tls/tls.html Is there any idea or a better well-explained user guide about tls support? Regards, Burak

2 1

Zero left padding IP in Kamailio
by Techie Sup 30 Dec '12

30 Dec '12

Hello, My IP is 192.168.1.25 and I have added left zero padding i,e: 192.168.001.025 in the configuration file and try to start kamailio, it works.. but along with that rtpproxy does not start. Once I revert back it works. My script requires to add zero left padding for a specific purpose and for that I cannot remove the left zero padding. Any solution to get this work? Regards Selvam

2 1

kamailio-ser merging finished
by Daniel-Constantin Mierla 29 Dec '12

29 Dec '12

Hello, in November 2008 was decided to start integration of kamailio and ser project, to get to same source code. About one year later, there was a single source code tree, but there were several duplicated modules that kept some confusion around the two projects. As the time passed, it was shown better what modules are in the focus, making the decisions to sort them out easier. The past weeks were dedicated to removing duplicated, process that just ended -- at this moment there is no duplicated module, all of them being integrated. In summary, the modules from ser that had no conflict were moved as is to modules folder, eventually after rename. The modules with conflicts were merged, adding missing features to the most developed ones, moving the others to obsolete folder. It happened that kamailio version of the modules stayed in the majority of the cases, because they had larger set of features (the core in v3.0.0 was kept from ser). Please note that the extra features from ser modules might not be in the same form, but alternatives that should offer same functionality at the end. People using ser flavour so far should start testing the master branch. Do ask questions on the mailing lists if something that you used to have in config is not longer available, so we can assist in migration and, hopefully, create a migration guide on wiki. There might few bits and pieces to tune here and there, soon we will get into code freezing period for testing the next major release, a good time for such adjustments. It is important that everyone using former ser modules to report the issues that are found. Many thanks to all those involved in integration efforts as well as in new developments that kept the project flying! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

GROUP - change database table
by Daniel W. Graham 28 Dec '12

28 Dec '12

I am attempting to move the location of the GROUP 'grp' table. I created column grp in subscriber table and configured column with same settings as default table and updated column name as follows: modparam("group", "db_url", DBURL) modparam("group", "table", "subscriber") modparam("group", "use_domain", 1) As the other required column names are present in the subscriber table and also default params for this module, I didn't specify them, but I have also tried doing that with same result. I have debugging enabled and when attempting to restart / start kamailio it simply doesn't start and no debug errors are detected, also nothing in log file. If I change to modparam("group", "table", "grp"), kamailio starts fine. Any ideas out there as to what would cause this? Thanks for your time- -dan [cid:image001.gif@01CDDEC4.BE0AD670]

2 4

Is somebody at 29C3 in Hamburg and wants to meet for a drink and some VoIP gossip?
by Klaus Darilion 28 Dec '12

28 Dec '12

regards Klaus

1 0

No Conect
by Matheus Perillo - Rede Itatiaia 27 Dec '12

27 Dec '12

Good afternoon , My name is Matthew and Perillo on IPtel have an account with the username of djperillo . I am not able to connect with anyone. We have to work in the company of various equipment and all with Comrex IPtel account and you can not connect with any of them . Everything indicates to the page Comrex was made and does not connect . The Access are with the SIP Status On LINE . On my iphone the phone symbol on Media5 is green but when I try to connect with the Comrex Access the following message appears " user / password denied , please check your settings and try again . " What can I do to solve this problem ? Thank you . Matheus Perillo Técnico de Manutenção Tel +55 31 2105.3545 Fax +55 31 2105.3677 matheusperillo(a)itatiaia.com.br www.itatiaia.com.br Rua Itatiaia 117 • Bonfim • Belo Horizonte • MG • Brasil • 31210-170

1 0

retransmitted INVITE's
by schwatzi＠web.de 27 Dec '12

27 Dec '12

2 1

extension of SNMPStats module
by 马天 27 Dec '12

27 Dec '12

Hi all, Merry X'mas. I'm working on a project which uses kamailio as sip server. Since I'm really new to kamailio and sip protocool and c, I need some help.My question is that is it possible to extend snmpstats module so the snmpstats module can respond to customized oid.I already wrote the mib file and generated the corresponding c files which handle the request. But I couldn't compile it into snmpstats module.Always get a "No Such Object available on this agent at this OID" when i send GET request using snmpget.Thanks. Regards, Sebastian

2 1

OPTIONS packet from dispatcher module
by Morten Isaksen 27 Dec '12

27 Dec '12

Hi, I have an issue with the OPTIONS packets from the dispatcher module does not contain a max-forward header. This causes the gateway in the other end to not reply to the OPTIONS packet. Can anyone please give a hint where en the code I need to change that. I noticed that dispatcher module uses the tm module to send the OPTIONS but after that I got lost in the code. -- Morten Isaksen

2 1

Problem with RLS
by Dmytro Bogovych 26 Dec '12

26 Dec '12

Hi. I still try to setup RLS capable kamailio instance. Now the server accepts my services and resource list documents. It generated backend subscriptions - i can see it in pua table. When another peer publish its presence - presence tries to send notify to sip address specified in modparam("rls", "server_address", " sip:rls@sip.voipobjects.com"). But the notify does not come back to rls - it is rejected by somewhere in core or other module with error 404. I tried to create the rls(a)sip.voipobjects.com user - it did not help. May anyone give me advice?

1 1

Presence rules?
by Dmytro Bogovych 25 Dec '12

25 Dec '12

Greetings. I try to add rls/xcap presence into resiprocate-based user agent. May anyone tell me what module is responsible for parsing presence rules documents? I'm trying to setup "allow all" presence rules document but did not get success... Thank you!

2 4

Merry Christmas!!!
by Raj Roy Ghandhi 25 Dec '12

25 Dec '12

I WISH Merry Christmas!!! to my Kamailio FAMILY !!!! Best Regards, Roy.

1 0

obsoleting modules_s/*radius modules
by Daniel-Constantin Mierla 24 Dec '12

24 Dec '12

Hello, according to the archive, modules_s/*radius modules were kept because of duplicate and incompatible auth modules at the time there was an attempt to remove them: - http://lists.sip-router.org/pipermail/sr-users/2010-June/064318.html Now there is a single auth modules, thus I guess the modules_s/*radius can be moved to obsolete. If anyone has different opinion, reply with the arguments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 15

Queries
by Ankit sharma 24 Dec '12

24 Dec '12

Hi, I have newly installed Siremis sip router. But I have no idea How I could start or How do I start calls? If you can provide me any Manual so it would be great. Sincerely Ankit

4 3

How to set variable externally
by Mino Haluz 22 Dec '12

22 Dec '12

Hi, I would like to set my custom different debug levels (with flag?) externally with kamctl command. So I neednt restart kamailio if I want to enable/disable debug. Which module should I use in that case? Thanks, Mino

6 12

AUTOREPLY Re: Problemdebian squeze uac module load failed
by strobel＠hfo-telecom.de 21 Dec '12

21 Dec '12

******************* AUTOREPLY VON miconda(a)gmail.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 27.12.2012 bis einschließlich 01.01.2013 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Brunner oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hello, do you get another error message before? If not, set debug=3 in kamailio.cfg and run again. Then look at the logs to see if there is another message giving hints about what is wrong. Probably some mis-configured parameter. Cheers, Daniel On 12/21/12 4:38 PM, Thomas Hoellriegel wrote: > Hi all, i using debian squeeze. > I installled kamailio from the package system. > I set in the kamailio.cfg file: > loadmodule "uac.so" > I become the error: > Dec 21 16:37:35 1a-7567 > /usr/sbin/kamailio[3015]: ERROR: <core> [sr_module.c:939]: init_mod(): > Error while initializing module uac > (/usr/lib64/kamailio/modules_k/uac.so) > > Can you help plese? > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste für blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

RPMs for v3.3.3
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, for the records, the RPMs for v3.3.3 are now available (32b and 64b: CentOS 5 and 6, RedHat 5.0 and 6.0, OpenSUSE 11.4 and 12.1, Fedora 15 and 16). They are hosted by opensuse build service, links specific per distribution are listed at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] [sr-dev] kamailio-ser integration - db modules
by Ovidiu Sas 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:20 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> What's the difference between the db1 and db2 interface? > > It is not the case here, modules_k/ldap does not have any relation to > database interface. Any database module db_[DatabaseType] implements an API. SER was using one API and opensips/kamailio a different API (and therefore today we have version 1 and 2). All db modules in the stable release are supporting both APIs. The new db ldap module has support only for the SER API version. The API is documented in each lib/srdb[1|]/*.h header files. The ldap module from ser is completely different than ldap module from kamailio (despite the fact that both are using the same name). It's the same with the dialog module. Check the README file for each. Hope this brings a little bit of light :) -ovidiu

1 0

Re: [SR-Users] [sr-dev] kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

On 12/20/12 4:13 PM, Ovidiu Sas wrote: > On Thu, Dec 20, 2012 at 9:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >>>> - nathelper - some extra functionality, not sure if can be kept >>>> completely >>> Maybe Andreas can look into this, as there is a lot of work going on with >>> nathelper and the new rtpproxy anyways. >> I think Ovidiu Sas looked at it when he split the rtpproxy out in a >> dedicated module. > IIRC, ping_contact whas the extra functionality in nathelper: > http://sip-router.org/docbook/sip-router/branch/master/modules_s/nathelper/… > I don't know how widely used is this functionality. > > Maybe we should have a separate thread per module (in user mailing > list to gather more imput) and see if it's worth merging the code or > use only the k version. I re-cc-ed the thread to users in case someone has comments to it. > > Also, on a separate note, I saw the we have a few db2_[module]. > I think it would make sense for these modules to rename them into > [module]_db[1|2]. > For example: ldap - we should have both versions under modules: > - ldap_db1 > - ldap_db2 > Just a suggestion ... The type of the two ldap modules are different, modules_k/ldap is a connector to ldap server from configuration file, offering possibility to do ldap search queries from config. The former modules_s/ldap (now db2_ldap) is a DB API v2 implementation driver module, so it can be used as a replacement for db_mysql (for example) when using some modules (such as db2_ops). I prefixed with db2_ to indicate that is not implementing DB API v1. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, most of the duplicated or ser-specific modules were sorted out at this time, next is the situation with the remaining ones. A) modules that will be renamed using a prefix 'uid_', because they have a database schema using unique id instead of username and domain: - auth_db => uid_auth_db - avp_db => uid_avp_db - domain => uid_domain - gflags => uid_gflags (not documented, but this module loads so called ser global avps) - uri_db => uid_uri_db B) modules to become obsolete with some work to be done - cpl-c - seems that giving a parameter with what user to be used for loading the cpl scripts will make k and s versions compatible - maxfwd - different names for config functions - nathelper - some extra functionality, not sure if can be kept completely - textops - seems to have some extra cfg functions - pike - has a rpc command - registrar/usrloc - avps stored per contact C) modules to become obsolete, most probably without further work - permissions - different structures/database table internally, but all should be achieved using k version (data migration might be required) D) modules that needs further investigations - rr - there seems to be some feature related to avps and record-route parameters/cookies - this module could be just renamed at the end (like rrs) E) Sample modules, not for real use - they can be kept in this directory (modules_s[ample]) - print - print_lib If anyone has comments regarding these remarks, reply here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Bug in uac_replace_from if used more than once?
by Daniel Tryba 20 Dec '12

20 Dec '12

Apparantly it should be simple to mangle the From in request: http://kamailio.org/docs/modules/stable/modules_k/uac.html#id2525538 The task at hand is use the prefix 131 to suppress callerid for this call: if($rU=~"^131[0-9]+$") { strip(3); append_hf("Privacy: id\r\n"); uac_replace_from("anonymous","sip:anonymous@anonymous.invalid:$rp"); uac_replace_to("","sip:$rU@$rd"); } Just before this replace there is an other uac_replace_from to rewrite an alphanumeric username to the default callerid for that device. If there are 2 uac_replace_from in the path, things go wrong like below. A uac_restore_from() before the second uac_replace_from doesn't change this behavior. device calls 1310123456789: U 10.0.34.226:1300 -> 10.0.35.226:5060 INVITE sip:1310123456789@kamailio.local:5060 SIP/2.0. From: <sip:anonymous@10.0.3.44:5060>;tag=26e5ce5850. To: <sip:1310123456789@kamailio.local:5060>. Kamailio forwards request to dispatcher but fails to properly replace the from uri: U 10.0.35.226:5060 -> 10.0.32.36:5060 INVITE sip:+31123456789@kamailio.local:5060 SIP/2.0. From: "anonymous" <sip: +31987654321@10.0.3.44:5060sip:anonymous@anonymous.invalid:5060>;tag=26e5ce5850. To: <sip:0123456789@kamailio.local>. Even though the debug log suggests it did the proper replacement: uac [replace.c:265]: value to store is is '<sip:anonymous@10.0.3.44:5060>' and len is '30' uac [replace.c:268]: Storing in FROM-AVP (for use in reply): '<sip:anonymous@10.0.3.44:5060>' with len '30' uac [replace.c:324]: uri to replace [sip:anonymous@10.0.3.44:5060] uac [replace.c:325]: replacement uri is [sip:anonymous@anonymous.invalid:5060] uac [replace.c:383]: encode is=<AAAAAAAAAAAAAAAAAABQXkFeV15BQUcUXF5AUWxpZDo1MDYw> len=48 uac [replace.c:265]: value to store is is '<sip:1310619024127@kamailio.local:5060>' and len is '54' uac [replace.c:271]: Storing in TO-AVP (for use in reply): '<sip:1310123456789@kamailio.local:5060>' with len '54' uac [replace.c:324]: uri to replace [sip:1310123456789@kamailio.local:5060] uac [replace.c:325]: replacement uri is [sip:0123456789@kamailio.local] uac [replace.c:383]: encode is=<AAAAAAEFAAkGAw0BAANxQlYuERJDCRxeHAoXWgYRXRdbAxtNHxwNCgdACxgubmw6NTA2MA--> len=72 Kamailio version: kamailio 3.3.2 (x86_64/linux) (the latest stable debian/squeeze package). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

3 5

Changing METHOD
by Olle E. Johansson 20 Dec '12

20 Dec '12

Hi! I want to change METHOD on a SIP message from UPDATE to INVITE or REGISTER to INFO. Seems like $rm is read-only. Any other known way to change a requests method? /O

4 8

Send reply with Reason header
by Mino Haluz 19 Dec '12

19 Dec '12

Hi, I know how to use sl_send_reply, but I would like to add "Reason:" header to the reply generated. Is there any command or variable for that? Thank you, Mino

3 3

Video conference with many users
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, I want to do web (HTML5 + WebRTC) Sip client which can do the video conference with multiple users. Current release of SIPML <http://www.sipml5.org/> does 1 to 1 call. I have no idea of conference with many users. Is it the client that we need to modify to accept call and join the conference ? Do I need to send INVITE with extra parameters ? Please advice me. Best Regards, Roy.

3 2

Question registering sip providers in kamailio and forward all?
by Thomas Hoellriegel 19 Dec '12

19 Dec '12

Hi all, I.m. a newbi in Kamailio. My problem: I.m. a blind user and i programmed a voicechat in Freeswitch. 3 have 2 sip providers. vodafone.de sipgate.at and freenet.de. The problem: i must register these proviers in kamailio. And forward all incoming calls to Freeswitch, and outgoing calls from freeswitch to kamailio. Dtmf modes are: auto rfc2833 and inband. Then i call: dorf(a)dorf.blindi.net i like to forward these calls to my freeswitch box. the kamailio ip ist: 217.172.180.108 and the freeswitchbox: 217.172.170.120 Can your help me please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

2 1

NAT Traversal issue
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client->LAN client web page call 2. LAN webrtc client -> 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy.

2 2

New developer: Richard Good
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Richard Good - he is among those developing the IMS extensions for Kamailio (http://code.google.com/p/smile-ims-extensions/), to be merged in the next days to the GIT master branch. His git commit id is: richard.good My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 2

modules_s/timer review
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I was comparing modules_s/timer with modules_k/rtimer to spot the differences. modules_s/timer is working only with core timers (fast and slow timers), while the modules_k/rtimer can start own timer processes, or register a task to main core timer. Other differences ... modules_s/timer: - run on milisecond basis - option to enable/disable timer modules_k/rtimer - run either on second or micro-second basis - option to execute the many route blocks on same timer So, apart of enable/disable timer option, rtimer can do more than what timer module offer (mili-seconds can be run as 1000x micro-second). Enabling/disabling a timer can be workarounded from script, using a shared variable tested at begininning of rtimer route, like: route[RTIMER] { if($shv(rtimer)==0) return; ... } Enabling will mean setting $shv(rtimer)=1, either from other parts of the config or via MI/RPC commands. Now, both modules can be kept, no naming conflict, it is just a matter of code maintenance if we should keep timer module. Enabling/disabling timers can be added to rtimer in the future, but it is unlikely to happen before next major release (unless someone else does it). Is anyone here using modules_s/timer module aware of other differences comparing with rtimer? Looking for comments to keep or obsolete modules_s/timer module ... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.3 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning next minor release - v3.3.3
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

dlg_manage() and add_contact_alias() problem
by Pavel Miskov 18 Dec '12

18 Dec '12

Hi all, I want to end dialog (Kamailio sends BYE to both parties) that is lasting longer then some defined time. everything is working fine when using old fix_nated_contact function, but when using new add_contact_alias function kamailio sends BYE to private addresses. 1. "kamctl fifo dlg_list" output when using fix_nated_contact(): caller_contact:: sip:bob@PUBLIC_IP:50784 callee_contact:: sip:alice@PUBLIC_IP:64472 2. "kamctl fifo dlg_list" output when using add_contact_alias(): caller_contact:: sip:bob@10.2.5.206:10000 callee_contact:: sip:alice@10.2.5.205:20000 dlg_manag() doesn't store contact alias for caller and callee so Kamailio is sending BYE to private addresses. Am I doing something wrong or I can't use possibility of ending dialog with contact aliases? Thank you Pavel

2 3

Support
by Murali Krish 18 Dec '12

18 Dec '12

Hi this is Murali krishnan i am final student of master degree in Dalhousie University, I am Working on my project, i m need of a sip server which can support H.264 codec video calls...I created an account in iptel sip account when i used the account it doesn't support video calls. If i use SER will this support video calls? -- *Regards Murali Krish*

2 1

From sip phone to provider trunk
by andre second 18 Dec '12

18 Dec '12

Hi,I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that?Thank you!

2 4

p_usrloc registration update problem
by Hans Goossens 18 Dec '12

18 Dec '12

Hi, I'm testing the p_usrloc module (Kamailio version 3.3.2, installed via ubuntu package) and it looks like it's working but for some reason a re-registration is causing a new entry being added to the distributed location table instead of updating the existing one. The kamailio configuration looks like this: #!define DBURL "mysql://openser:openserrw@localhost/openser" loadmodule "p_usrloc.so" modparam("p_usrloc", "db_mode", 3) modparam("p_usrloc", "write_db_url", DBURL) modparam("p_usrloc", "read_db_url", DBURL) modparam("p_usrloc", "write_on_db", 1) modparam("p_usrloc", "alg_location", 1) modparam("p_usrloc", "reg_db_table", "locdb") mysql> select * from locdb; +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | id | no | url | status | errors | failover | spare | rg | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | 1 | 1 | mysql://root:secret@172.16.10.131/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | | 1 | 2 | mysql://root:secret@172.16.10.132/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ When using the default usrloc functionality everything is working fine. The only difference I noticed is that the default usrloc is using MYSQL UPDATE to update the location entry and that p_usrloc is using MYSQL REPLACE. Regards Hans

1 0

variable in rewritehost ?`
by Ali Jawad 17 Dec '12

17 Dec '12

Hi I am using slqops to get some user related info from location table, I want to use that info in rewritehost, however when I do use a variable in rewritehost I get an error "Unable to resolve variable". Is it possible to use a variable in rewritehost ? If not what is the workaround ? Thanks

3 9

Dialog module not decrementing profile size
by Grant Bagdasarian 17 Dec '12

17 Dec '12

Hello Community, I've recently started to use the Dialog module to manage the amount of incoming calls from our SIP provider allowed to pass through by using profiles_with_values and setting the size of these profiles to a certain amount. Most of the time this works as it should, but sometimes the 'slots' available for a profile aren't cleared even though a dialog has finished. This seems to happen when an high amount of regular traffic is hitting kamailio. Has anyone else experienced this? Are there perhaps any other ways to control the slots available for certain incoming SIP traffic? Below parts of my configuration file. Perhaps I'm doing something wrong which causes this to happen. In my route[WITHINDLG] -> loose_route -> is_method("BYE") I'm not calling dlg_manage(), could this be the problem? Other comments are welcome too, as I'm fairly new to Kamailio. modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value", "318005004"); modparam("dialog", "default_timeout", 43200) modparam("dialog", "dlg_match_mode",2) modparam("dialog", "detect_spirals", 1) request_route { # per request initial checks route(REQINIT); # 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; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE")) { xlog("L_INFO", "Incoming request for $rU from $fU"); if ($rU == "318005004") { route(DIALOG); } } # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[DIALOG] { dlg_manage(); record_route(); $var(SIZE) = 0; sql_query("vf", "exec dbo.SelectAvailableSlots '00$rU'", "ra"); $avp(s:limit) = $dbr(ra=>[0,7]); xlog("L_INFO", "Maximum simultaneous calls is configured at $avp(s:limit) ...."); sql_result_free("ra"); if ($rU == "318005004") { get_profile_size("318005004", "$rU", "$var(SIZE)"); } xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, excluding the current call"); if($var(SIZE) >= $avp(s:limit)) { xlog("L_INFO", "Simultaneous calls limit of $avp(s:limit) reached for $rU: $var(SIZE)\n"); sl_send_reply("603", "Simultaneous calls limit reached"); exit; } if ($rU == "318005004") { set_dlg_profile("318005004", "$rU"); } if ($rU == "318005004") { if(get_profile_size("318005004", "$var(SIZE)")) { xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, including the current call"); } } } route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #Used for call transfer if($fU == "318005004") { if (is_method("UPDATE")) { sl_send_reply("405","Method not allowed"); xlog("L_INFO", "Dropping UPDATE message...\n"); exit; } if (is_method("INVITE")) { record_route(); } if (is_method("BYE")) { dlg_manage(); } } if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... 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() ) { # non loose-route, but stateful ACK; # must be 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; } } Regards, Grant

2 8

status of kamailio stun server?
by Juha Heinanen 16 Dec '12

16 Dec '12

i tried to use kamailio built-in stun server, but failed, because it supports only binding requests. my ua tried to send it "allocate request lifetime" request, which failed with 600 global failure response. what is the status of kamailio's stun server? is anyone interested in developing it further or should we obsolete the server and start using some external server? -- juha

2 5

Writing User-preferences Username into Radacct Table on Call Forward
by Jonathan Hunter 15 Dec '12

15 Dec '12

Hi All, I was wondering if anyone would be kind enough to let me know if they have implemented anything like as follows in Kamailio? We are looking to implement call fowarding, activating it as usual in usr_preferences table. I was wondering if on the outbound leg its possible to write the username from usr_preferences with CFW active into the radacct table easily? This is with Radius and opensips DBs in Mysql. Many thanks Jon

2 2

NAT Traversal :: RTPProxy and Jitsi client
by Raj Roy Ghandhi 14 Dec '12

14 Dec '12

Hi, I am trying to communicate 2 Jitsi clients in 2 separate private networks. Both clients are behind NAT. Kamailio Server is on Public IP. Both Jitsi clients does register and presence works fine. And also text messages works well. But unable to call each other. Had some one had the same issue before. I tried with many soft clients and all failed with same result. Same scenario works with IP Phones. Can anybody point out what I miss there. Regards, Roy.

2 2

OT: MSRP over WebSocket Javascript stack
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi, Crocodile has just open-sourced our MSRP over WebSocket (see http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/ The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 2

RTP Proxy Interaction
by Austin Einter 14 Dec '12

14 Dec '12

Dear All I am looking at Kamailio's module, file, function that receives rtp and rtcp port from rtp proxy. Can somebody please let me know which file/function I need to check. Thanks Austin

2 1

Re: [SR-Users] Help
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi Roy, Please keep emails on list. Regards, Peter On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote: > Dear Peter, > > Thanks for all the support that you gave me during the Kamalio > configuration with RTPProxy. > Currently it works fine with IP Phones but not soft clients. (Jitsi) > > > If I give you my server information would you be able to check that > out for me. > > > I was lost for that issue. > > > Best Regards, > Roy. -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Local call routing accross many kamailio servers
by Ali Jawad 14 Dec '12

14 Dec '12

Hi I do have 3 kamailio servers sharing the same db, calls go to Asterisk and that works great, I did purchase a DID and point it to one of the kamailio servers, it works fine so far for that kamailio server as the call is forwarded to the softphone ..but ONLY if the user is registered on the kamailio server the DID is forwarded to..if the kamailio user is on another kamailio server it does not work..although the database and location tables are shared. Simple diagram DID Provider <=== DID Kamailio ===> Kamailio Servers A, B, C If a softphone registers on DID Kamailio and I call the DID it works, if a softphone registers on Kamailio A or B or C, I never get the call, looking at ngrep the call just stays on DID Kamailio and it does not try to send it to any other server, how can I make the Kamailio DID lookup the destination from the location table and send the call there ? Regards

2 9

Hardware requirements voip servers
by Grant Bagdasarian 14 Dec '12

14 Dec '12

Hello, We are considering upgrading our VoIP platform and I'm wondering what kind of hardware we need for our requirements. The architecture will consist of a SIP Proxy Server (Kamailio) in front and a cluster of Media/Application Servers behind it. The requirements are based on a single server. SIP Proxy Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. - RTP Proxy - Record-Route Media/Application Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. What do we need in terms of CPU, Memory, HDD when looking at the following specific requirements: - 100 CPS - 200 CPS - 300 CPS Which hardware components are critical for handling this many calls, is it CPU, Memory, Disk IO bound? Also which open source media/application server software is suitable for these requirements? We are currently using Asterisk, which works great, but perhaps there are others available which are more suitable for these requirements. Thanks in advance! Regards, Grant Bagdasarian

1 0

Kamailio <-> Asterisk MWI
by Daniel Tryba 13 Dec '12

13 Dec '12

I'm trying to get the MWI indicator on endpoints to work by redirecting the SUBSCRIBE to the voicemail server (asterisk) using usc_replace_(to|from). This works fine for the initial request but the next subscribe will fail since the to/from don't get rewritten, which is indicated by the error: ERROR: uac [replace.c:250]: decline FROM replacing in sequential request in auto mode (has TO tag) This occurs when the modparam "uac","restore_mode" is manual or auto. When set to none the redirect will be correct but all responses have to be rewritten manually (I'm a beginner and I don't want to do this just yet). And since uac has knowledge about this having being rewritten, sending the original from/to looks like like a bug to me (more probable a misconfiguration/omission of something crucial withing dialog responses :) Changes to the (default debian package) config, in route route[PRESENCE] in if(is_method("SUBSCRIBE")) if( is_method("SUBSCRIBE")) { if(search_hf("Event", "message-summary", "a")) { if (!www_authenticate("$fd", "subscriber")) { www_challenge("$fd", "1"); exit; } if(avp_db_query("SELECT value FROM usr_preferences WHERE username='$au' and attribute='voicemail' ORDER BY value limit 1")) { $avp(voicemail)=$avp(i:1); $avp(subscrfrom)=$fu; avp_subst("$avp(subscrfrom)", "/sip:.*@/sip:$avp(voicemail)@/"); uac_replace_from("$avp(subscrfrom)"); uac_replace_to("","sip:$avp(voicemail)@$sel(cfg_get.voicemail.srv_ip): $sel(cfg_get.voicemail.srv_port)"); remove_hf("Authorization"); $ru = "sip:" + $avp(voicemail) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); } } else { handle_subscribe(); t_release(); } } exit; 192.168.34.226 is the sip device, 192.168.32.40 kamailio, 192.168.32.41 the voicemailserver. Initial request: U 192.168.34.226:5852 -> 192.168.32.40:5060 SUBSCRIBE sip:grandstream@sip.local SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>. Gets rewritten to: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:0880100782@sip.local>;tag=67d8f67065827fe2. To: <sip:0880100782@192.168.32.41:5060>. The response gets relayed to the endpoint and MWI will light up since there is a message. But next subscribe update from the endpoint triggers the "decline FROM replacing in sequential request" error and thus the subscribe send to voicemail is the original to/from: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>;tag=as324de8f3. Resulting in a 404 since the to/from are not the real peername on the voicemail server. How do I force Kamailio to rewrite the from/to without doing this manually for a requests/responses (restore_mode none)? The easy way out of this problem would be to have a 1 to 1 relation for the Kamailio subscribers and voicemail peers. But uac_replace should work (somehow). Attached is a full capture of the subscribe/notify. Running 3.3.2+squeeze from http://deb.kamailio.org/kamailio/. Machine is also running topoh module (192.168.0.1). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 6

NAT for Load Balancer and Proxies
by Nguyen Anh Tuan 13 Dec '12

13 Dec '12

Hi everyone, I have 1 Load Balancer and 2 SIP Servers, all of them are Kamailio. Does anyone suggest how to add NAT for them? Best Regard, Nguyen Anh Tuan

2 1

SIP proxy changing Contact header of presence server requests
by Owen Lynch 13 Dec '12

13 Dec '12

Hi, we have 2 separate instances of kamailio running as a SIP proxy and presence server. The proxy uses the nathelper module to correctly change the Contact headers of client requests. However it also does the same for notify requests from the presence server, which results in incorrect RURI from subsequent client subscribes. The NATDETECT route is shown below, I plan to leave the existing nat_uac_test but also explicitly skip the fix_nated...() calls for requests from the presence server. Does this seem reasonable? # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } Thanks, Owen Lynch

2 1

Dispatching to Asterisk cluster
by Grant Bagdasarian 12 Dec '12

12 Dec '12

Hello, We have a cluster of three Asterisk machines. Each machine answers an incoming call and transfers it to unique remote destination. So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02, Asterisk03 to Destination03. The Asterisk machines are allowed to receive and answer only 50 calls each. I have a Kamailio in front of these three machine. It is currently used to dispatch calls to the three machines using round-robin. But I need some way to equally dispatch a maximum of 50 calls to each device in the Asterisk dispatcher set. I was thinking of using the dispatcher module and using the 10 algorithm, to load balance based on call load. But I'm not sure if that will guarantee a maximum of 50 calls being dispatched to each Asterisk. Theoretically, if I set the ds_hash_size to 7, which is 128 slots, at some point two of them should have 43 calls and one 42 calls. Also the value of ds_hash_size should be the power of two. Are decimal values allowed, like 7,229 which equals to almost 150? Perhaps I should use the ratelimit module for this? I'd appreciate it if someone could give me some advice on this. Regards, Grant

2 2

Syntax of xavp
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, I'm playing around with xavp, but there are some things I can't wrap my head around. What basically works is this: $xavp(a=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; Getting the value like this: xlog("L_INFO", "a-foo='$xavp(a=>foo)' and a-bar='$xavp(a=>bar)'"); as well as xlog("L_INFO", "a-foo='$xavp(a[0]=>foo)' and a-bar='$xavp(a[0]=>bar)'"); ... works, because when omitting the index, it assumes [0]. I really like to avoid such implicit assumptions though, to make the config really clear, so I prefer the second approach. What doesn't work for me then is explicit assignment of the very first value, like this: $xavp(a[0]=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; If I do this, the "a" xavp is never created. Wouldn't it be good practice to create the xavp on first assignment? That way, I don't have to take care in the config file whether the xavp is used for the first time, where I have to omit the index. What I'm also wondering is whether it's possible to directly access nested xavps, like this: $xavp(a=>foo) = 'afoo'; $xavp(b=>foo) = 'bfoo'; $xavp(c=>a) = $xavp(a); $xavp(c=>b) = $xavp(b); xlog("L_INFO", "a-foo='$xavp(c=>a[0]=>foo)'"); or xlog("L_INFO", "a-foo='$xavp(c=>a=>foo)'"); Both ways give me "a-foo='<<xavp:0x7f0d387fe178>>'". Is this even intended? Would be really cool if that's possible! Thanks, Andreas

3 4

NAPTR, SRV and sips vs. transport=tls
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, Hope to get some guidance here over the usage of "sips" and "sip plus transport=tls" when following RFC3263. The RFC3263 says that a NATPR record could return something like this for a query like "host -t NAPTR example.com": ; order pref flags service regexp replacement IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.example.com. IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.example.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com. This means that the client should use sips if possible when contacting example.com. Concluding from that, I suppose it should perform an SRV lookup "host -t SRV _sips._tcp.example.com", and the result would be: ;; Priority Weight Port Target IN SRV 0 1 5061 server1.example.com IN SRV 0 2 5061 server2.example.com What I'm curious about is how requests towards one of these servers should look like, and how they are being handled by kamailio. A lot of clients and servers are sending "sip:user@example.com;transport=tls" in request URIs and Contact headers and Record-Route headers, and you can check with uri_param("transport","tls") which transport socket to use. This is pretty useful as you can determine hop-by-hop whether or not to use TLS. This approach has been obsoleted by RFC3261 though, and there doesn't seem to be a mechanism in RFC3263 to indicate "use schema sip, but use transport=tls". On encounter of a NAPTR record like the one above, how does kamailio act? Does it set a "sips" schema for the next hop? And what's the general take on this "sips" schema? As far as I understand RFC3261, it means that if a client sends a request to a sips-URI, the request is sent to the domain via TLS, and from there "the request is sent securely to the callee, but with security mechanisms that depend on the policy of the domain of the callee." (RFC3261, Chapter 4). What does this really mean in practice? Are you allowed to rewrite the schema to "sip" and pass it on for example via UDP to the callee if the callee didn't indicate transport=tls (deprecated anyways) or "sips:" in the Contact of the registration? Or should you keep "sips" as schema, but still send it via UDP, because you know based on local policy or based on client registration that the next hop is not supporting TLS? How would widespread clients react when getting a call to a "sips" URI, especially if they receive it via UDP? Looking forward hearing your input on that, Andreas

5 14

MySql error
by Hermes F. 11 Dec '12

11 Dec '12

I'm trying to create MySql database using the following /usr/local/sbin/kamdbctl create I followed the instructions on http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x but I receive the following error message: "ERROR: database engine not specified, please setup one in the config script" Can someone help me please ?

2 1

Regarding Kamailio Registrar
by AMIT SHAH 11 Dec '12

11 Dec '12

Dear all, We are using Kamailio SIP Server in our application. We want to know 1 thing: We are using Kamailio SIP server as registrar and proxy server. For registration database, we are using dbtext module. We maintain record of SIP Extensions who have registered in Our proprietary file too. Now when we boot the Kamailio SIP server, At that time our file should be empty i.e. previously registered Extension entries should be deleted or it must be kept as it is. But if during power off, any extension had sent REGISTER request to Unregister then it was not executed as kamailio SIP server was not ON. But at boot time, as its entry was present, Kamailio SIP server will Consider it a registered extension. Please guide us for this situation. Thank you, Amit Shah.

2 1

Unable to set custom RPID
by Mino Haluz 11 Dec '12

11 Dec '12

Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino

2 1

New developer: Konstantin Mosesov
by Daniel-Constantin Mierla 10 Dec '12

10 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Konstantin Mosesov - he has contributed patches to app_python and joins the team to help maintaining and developing this module. His git commit id is: ez My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

How to use LIMIT 1 in db_query api
by Krishna Kurapati 10 Dec '12

10 Dec '12

Hi, I need a way to limit the results to just one row. The only way I know I could do is to call db_query followed by fetch_result of 1 row. However this is not optimal as all the rows are fetched to dynamic memory with db_query. Is there an alternative approach? Thanks Krish Kura

2 1

Help
by ameneh pourmoghadas 07 Dec '12

07 Dec '12

Dear Admin, My name is Ameneh from India. I am new in SIP. I have installed asterisk 1.8 on my ubuntu system and could have registered the internal users in my LAN to talk over x-lite. Now my question is how can I make a call to a mobile in India or a PSTN number in another country that would require a public SIP I think. how can I make my SIP server as public? would you please help me? thanks and regards, Ameneh

7 6

CTL module and alternatives for remote execution mi commands.
by Pavel Zalunin 07 Dec '12

07 Dec '12

Hi. In our project we need to query current users, dialogs and ability to end dialog using some rpc. Requirements are - do this remotely. I already tried to use xmlrpc module. When we had 42 users online it returned 17kb broken xml file. Then I tried to use binrpc protocol from CTL module. Usage ends with returning broken message too (in case, when message is biger than 21kb). Are there any other more stable solutions to query mi information using some tcp protocol? P.S. When I discovered sources of CTL module I found usage of writev - here is the problem about partial packets, when buffer is longer, than 21kb it don't pushes all vector buffers into socket. And in general, this module is using async sockets, AFAIK writev don't gives any guarantees, that it will write all vectors in async sockets. P.P.S. Of couse, last resort solution will be writing proxy for using mi_fifo on kamailio side to forward all records to other servers, but if there are more pretty solutions, please give me know:) Thanks. Pavel.

2 2

OFFTOPIC - Realtime testers needed - Asterisk Path header support (Oolong branch)
by Olle E. Johansson 07 Dec '12

07 Dec '12

While this is not Kamailio - it's related. If you want to use Kamailio in front of Asterisk, you do need SIP Path header support in Asterisk to do it right. Especially if you have one Kamailio handling Internet-facing communication and one handling internal communication. Please test! /O Vidarebefordrat brev: > Från: "Olle E. Johansson" <oej(a)edvina.net> > Ämne: Realtime testers needed - Path header support (Oolong branch) > Datum: 7 december 2012 11:07:09 CET > Till: Asterisk Developers Mailing List <asterisk-dev(a)lists.digium.com> > Kopia: "Olle E. Johansson" <oej(a)edvina.net> > > Friends, > > I have taken Klaus Darillion's patch for Path header support in Asterisk and created branches for 1.4, 1.8 and trunk. I need help testing it with realtime support by someone who are more used to the realtime databases than I am. I also need help completing the LDAP schemas with the new attribute. > > General testing is also encouraged. > > I have been using this patch without realtime for a couple of months. So far no Squirrels where hurt in the progress. And no bugs hit me in the back. > > Read more about the patch here: > http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/R… > > Klaus' original review: > https://reviewboard.asterisk.org/r/991/ > > The SIP Path header support is documented in RFC 3327 (that has a strange title). > > Basically, this is used when you have a proxy between a phone and your Asterisk. Asterisk needs to remember the path to the device when sending an INVITE, so we save both the contact and the path (basically a route) to the phone when Asterisk receives a REGISTER. When sending an INVITE (or anything based on the registry) a route header is added and the message finds the proper way to the device. > > Path header support exists in the path module in Kamailio if you want to test. > http://kamailio.org/docs/modules/3.3.x/modules_k/path.html > > Looking forward to your feedback! > > /O

1 0

Kamailio\ Message 300 Contacts
by Fatima Chahrour~Vanrise Support 06 Dec '12

06 Dec '12

Dears, In the redirect message 300, is there any way to control the maximum number of contacts(IP's) returned in the message? (am only getting 5 IP's) Also, if I have one gateway with 5 IP's and trying the same call 3 times; the order of the IP's returned in the message 300 is different in each time. Why is that? Thanks in advance, F Chahrour

2 2

Kamailio 3.X.X + BLF configuration trouble
by Hemanshu Patel 06 Dec '12

06 Dec '12

Hello everyone, Till now we were working with asterisk to handle BLF. But then we decided to move everything to kamailio and hence i started working on BLF with kamailio. I have already configured kamailio for Call handling and presence information. >From different pages on Internet, I have found some information about configuration of kamailio+blf. And i did so, but somehow things are not working as they should. I have done following changes to kamailio.cfg file loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_mwi.so" loadmodule "dialog.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" modparam("presence", "fallback2db", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") modparam("dialog", "db_mode", 1) modparam("pua", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") Now when my phone(yealink VP530), sends subscribe request it looks as below: SUBSCRIBE sip:1001@172.16.27.66 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK10464096 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66> Call-ID: 1572901975(a)172.16.27.61 CSeq: 1 SUBSCRIBE Contact: <sip:1000@172.16.27.61:5063> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: VP530P 23.70.0.40 Expires: 120 Event: dialog Content-Length: 0 In response to this message, Kamailio returns 202 with following information SIP/2.0 202 OK Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK245273058 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8ee2 Call-ID: 1572901975(a)172.16.27.61 CSeq: 2 SUBSCRIBE Expires: 120 Contact: <sip:172.16.27.66:5060> Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Followed by a NOTIFY NOTIFY sip:1000@172.16.27.61:5063 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.66;branch=z9hG4bK3ccd.adfbc416.0 To: sip:1000@172.16.27.66;tag=2058189864 From: sip:1001@172.16.27.66;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6cd8 CSeq: 1 NOTIFY Call-ID: 2040123403(a)172.16.27.61 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:172.16.27.66:5060> Subscription-State: active;expires=170 During subscribe message asked for XML based reply, yet NOTIFY doesnt send any XML. and hence there is no impact of this message on device. Nothing is changed.Also there's nothing when i call from this device to another, ideally i should get some NOTIFY messages So, i think i am doing something wrong in Configuration. Can anyone please help me in it. If you need any other info, i'll provide you. -- Regards, Hemanshu Patel Senior Software Engg P Help the environment – please don't print this email unless you really need to!

3 7

Internal/External SIP setup
by Zero Aggression 06 Dec '12

06 Dec '12

Hello, I am a bit new to VoIP, and I just need a quick yes/no (and maybe pointers) answer. Is the following possible (and smart) to do with Kamailio? Kamailio server. Dual-homed (internal network, Internet). Internal users connect to Kamailio to make calls to internal and Internet SIP peers. Users have NO internet connection themselves (they can only reach the server). Can Kamaillio (when dual-homed) transport all the data/voice between Internet and intranet? Can it also be used to carry all traffic between intranet users (if there is no direct route between them)? Thanks so much. Zane

3 3

db_cluster configuration
by Juan José Ivars 06 Dec '12

06 Dec '12

Hello, i am configuring db_cluster module in kamailio 3.3 I have configured in roun-robin read between two MySQL servers but the load is not distributed equally My configuration is that: modparam("db_cluster", "connection", "con1=>mysql://USER1:PASS1@1.1.1.1/DB") modparam("db_cluster", "connection", "con2=>mysql://USER2:PASS2@2.2.2.2/DB") modparam("db_cluster", "cluster", "cls1=>con1=9r9p;con2=9r9p") And the auth module with: modparam("auth_db", "db_url", "cluster://cls1") But i have enabled the mysql logs and the major part of the querys are targeted to the second server Is there any cache by peer? What am i doing wrong? Thanks in advance.

2 1

Preparing for the next major release
by Olle E. Johansson 06 Dec '12

06 Dec '12

Hi! Daniel has set a code freeze date for the next major release and the Kamailio community needs to work together to prepare the release. That means you as well! There will be a lot of work in the coming weeks to merge code from SIP router into the Kamailio core to complete the merger process. We also have a large set of new modules and functions to test. The creation of a new release is much more than code - it's testing, documentation proof-reading and feedback. With a large and complex piece of software like Kamailio we can't test all functions with all possible combinations of configurations and connections to third party products - this is where YOU can help out! Read more on http://www.kamailio.org/w/2012/12/preparations-for-new-release/ and join us in creating the new release! /O

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 06 Dec '12

06 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 06 Dec '12

06 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Segmentation fault in sip-capture
by Owen Lynch 05 Dec '12

05 Dec '12

Hi, I am running Kamailio 3.3 as a Homer capture agent, the sip-capture module receives HEP packets from a sip-trace module in a remote kamailio instance. Periodically the capture agent dies. The log and core dump suggest the initial problem is an invalid HEP msg received, but kamailio crashes when trying to send an error reply. Can anyone shed any light on this? Thanks, Owen Lynch Back trace: Program terminated with signal 11, Segmentation fault. #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 3967 tcp_main.c: No such file or directory. in tcp_main.c Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.i686 keyutils-libs-1.4-4.el6.i686 krb5-libs-1.9-33.el6.i686 libcom_err-1.41.12-12.el6.i686 libselinux-2.0.94-5.3.el6.i686 mysql-libs-5.1.61-4.el6.i686 nss-softokn-freebl-3.12.9-11.el6.i686 openssl-1.0.0-20.el6_2.5.i686 zlib-1.2.3-27.el6.i686 (gdb) bt #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 #1 0x08136e78 in handle_tcpconn_ev (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4310 #2 handle_io (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4362 #3 0x0813ea3c in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x080a502b in main_loop () at main.c:1727 #6 0x080a6c00 in main (argc=11, argv=0xbfeafd14) at main.c:2546 Log: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: : <core> [pass_fd.c:288]: ERROR: receive_fd: recvmsg on 10 failed: Connection reset by peer Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: <core> [tcp_main.c:2347]: BUG: tcp_send: failed to get fd(receive_fd): Connection reset by peer (104) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sl [../../forward.h:193]: msg_send: ERROR: tcp_send failed Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=704 a=3 n=exit Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sipcapture [sipcapture.c:682]: ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[10] l:[10] vs [40] Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:638]: status: <500> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:640]: reason: <Invalid CSeq> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <yadayadayada>; state=6 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK2348.d1620f62.0>; state=16 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:785]: child process 19576 exited by a signal 11 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:788]: core was generated Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19575]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19574]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19573]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19572]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tls [tls_init.c:674]: tls module final tls destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:242]: shm_mem_destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:245]: destroying the shared memory lock Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [main.c:804]: terminating due to SIGCHLD

2 5

modules_s/eval - survey of usage
by Daniel-Constantin Mierla 05 Dec '12

05 Dec '12

Hello, is anyone using the modules_s/eval module? In the context of merging modules_s/* to modules/*, this one lacks the docbook xml files for readme. It has a text readme file, from where I understand it does script operations using a stack and polish notation expressions. Probably it was developed before the expressions support in config file language, at this moment all its operations can be done directly by the interpreter. I am considering moving the module to obsolete/ folder. If anyone is using it and wants to keep it in modules/*, then he/she has to convert the readme text into docbook format (which is not complex at all, just takes a bit of time, probably like 60 min or even less). If there is no answer in one week about usage, will be sent to obsolete/, from where it can be rescued in case of need (just to be clear for everybody that code is not lost). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

dns resolver issue (RFC3263)
by MÉSZÁROS Mihály 05 Dec '12

05 Dec '12

Hi All, I am experiencing an issue when i try to contact xy(a)cisco.com. I found that kamailio/sip-router can't resolve by default resolving way a TCP + SRV records from domain cisco.com. e.g. cisco.com misi@alma:~$ host -t NAPTR cisco.com cisco.com has no NAPTR record misi@alma:~$ host -t SRV _sip._udp.cisco.com Host _sip._udp.cisco.com not found: 3(NXDOMAIN) misi@alma:~$ host -t SRV _sip._tcp.cisco.com _sip._tcp.cisco.com has SRV record 1 0 5060 vcsgw.cisco.com. misi@alma:~$ host -t SRV _sips._tcp.cisco.com _sips._tcp.cisco.com has SRV record 1 0 5061 vcsgw.cisco.com. I can't call xy(a)cisco.com because it does not exists an NAPTR record in domain cisco.com, and furthermore no SRV with udp. But it exists sip+tcp record and also exists an secure sip SRV, so sips+tcp record! I read rfc3263 I find kamailio dns resolver is not working as it should according RFC3263. http://tools.ietf.org/html/rfc3263 _If no NAPTR records are found, the client constructs SRV queries for those transport protocols it supports, and does a query for each._ So kamailio should query all (udp, tcp, tls, sctp whatever) protocols. Can anyone help/guide me to create a fix to this issue? My plan is to create a patch to kamailio resolver, to correct and behave according RFC3263. Any help or guidance appreciated! Thanks, Misi

4 14

syslog message about invalid domain name
by Juha Heinanen 05 Dec '12

05 Dec '12

when i t_relay in-dialog bye: Request-Line: BYE sip:0x8eb94d0@[2002:c062:660a::1]:5050;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP [2002:C062:660A:0:0:0:0:1];branch=z9hG4bKde53.6aaed77ea908953bbb133e5021d74036.0 Via: SIP/2.0/UDP [::1]:5090;received=0:0:0:0:0:0:0:1;branch=z9hG4bKq~ouJaGs;rport=5090 From: <sip:jh@as.test.fi>;tag=5A29C828-50BEFE55000C7118-B4FF9B70 To: "" <sip:test@test.fi>;tag=92e26ffbfbc16e53 CSeq: 10 BYE Call-ID: 28d1e4872cc5f83e User-Agent: Sip Express Media Server (1.6.0 (x86/linux)) Max-Forwards: 69 Content-Length: 0 with $du set to sip:[2002:C062:660A:0:0:0:0:1]:59933;transport=tcp, i get to syslog: Dec 5 09:53:48 sip /usr/sbin/sip-proxy[1947]: INFO: <core> [resolve.c:729]: invalid domain name '[2002:C062:660A:0:0:0:0:1]' since everything is and works as expected, why is the info message produced and how can i get rid of it? -- juha

2 2

Xcap server chooice among openSER, mobicents and openXCAP
by zhi sun 05 Dec '12

05 Dec '12

hi guys, it is my first time to use XCAP, i want to implement XCAP server to support maybe huge amount of user accesses (as cloud solution), so far, i can find out three XCAP server solution: 1. openSER with XCAP Server Module http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html 2. mobicents with XCAP component/server build-in http://code.google.com/p/mobicents/wiki/Welcome?tm=6 http://www.mobicents.org/diameter/index.html http://hudson.jboss.org/hudson/view/Mobicents/job/Mobicents-Books/lastSucce… 3. openXCAP http://www.ag-projects.com/projects-products-96/101-openxcap but i am not clear to choose with solution to best match my situation (high volumn, high scale, and quick response). - does openSER support the cloud solution with high volum, high scale? - is it easy to use/integrate the XCAP server independently from mobicents? - is there performance issue for openXCAP, because it is implemented using python? any suggestions would be appreciated! best regards, -iamsy

1 0

io_wait errors
by Juha Heinanen 05 Dec '12

05 Dec '12

any idea what could be the cause for lots of these kind syslog messages: Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 56 Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 56 in the hash(-1, 0, 0xb2f28adc) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 59 in the hash(-1, 0, 0xb725a9d8) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 58 on 59: Broken pipe (32) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0xb3024a48 (flags 0x4018), fd 58 -- juha

2 2

[Kamailio-Users] BYE message not relayed to caller
by Vikram Ragukumar 04 Dec '12

04 Dec '12

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

3 9

openser_receiver
by Uri Shacked 04 Dec '12

04 Dec '12

Hi, I notice that under my /tmp folder there are openser_receiver filles as well (kamailio_fifo and kamailio_ctl are know....) Version is 3.3.2 latest branch. Any ideas? Thanks, Uri

1 0

Problem with Stateless Dispatcher Call Load Distribution (Algorithm 10)
by SIP Mailing-list 04 Dec '12

04 Dec '12

Hi guys, I wonder if someone could sanity check something for me. I currently have a simple load balancer set up using algorithm 10 (call load distribution) from the dispatcher module. I'm using Record-Routing so that I can clear down the call load record at the end of the call. Everything looked like it was working okay until I just spotted a problem that I think might be a bug. It's to do with ACKs to non-200 responses. In a normal call, the flow goes: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 200 OK --- ds_load_update() 4 <-- 200 OK 5 ACK --> 6 ACK --> Now between step 3 and 4 the proxy runs ds_load_update which according to the documentation "set internal state to confirmed for the call load" entry in the internal call state store. This means the proxy knows where to send the ACK at step 5 because it is safely in the call state store so you can use ds_select_dst again on the ACK to get it to the right place. Works fine! Now consider the non-200 case: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 603 Decline --- ds_load_unset() 4 <-- 603 Decline 5 ACK --> 6 ACK --> ??? Between step 3 and 4 the proxy is supposed to run ds_load_unset() which unconditionally removes the call from the call state store. Now when the ACK comes in at step 5, the proxy has no record of the call and so doesn't know where to send the ACK. This results in retried Declines and ACKs. Broken :-( If the proxy doesn't run ds_load_unset() on the 603 response, is there some kind of timer that will cause the call to be removed from the call state store on unconfirmed calls? I don't think I can run ds_load_unset on the ACK because there's nothing in the ACK that tells me it is because of a 603, rather than a 200. Edit: actually there is! The Route header is present on the 200 ACK, but * not* on the 3xx to 6xx ACK. Maybe I can use loose_route... Anyway, any thoughts would be appreciated! Cheers, Richard

2 4

default flavour set to kamailio
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, based on community interaction, development, package distributions and documentation maintenance in the recent years, the default flavour has been switched now to 'kamailio'. As written in my previous email, we started the last phase to complete the integration between kamailio-ser, aiming to get to a unique database schema and set of modules in maximum 6 months. Kamailio is the most spread and popular flavour, presented in many linux/unix distributions, being the one released and packaged periodically. So it will stay as main flavour, the one officially released by the project. Those that want can take the sources and build their own flavour, as the support in Makefile system is kept. Once the integration is finished, the flavour system will just set the application name, nothing else. From the open source project perspective, the resources will be allocated only on releasing under the main/default flavour (Kamailio). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Last phase of integration for kamailio - ser
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, we plan to start the last phase of integration between kamailio and ser. After about 4 years with same source code tree, there is still some overhead caused by several duplicated modules. Following the decision to go on with the most developed components and absorb the missing features from the duplicates, kamailio modules are those that will stay (move to modules/ eventually) and get updates when needed. We used same decision to keep the core and makefile system from ser, adding to it what was extra in kamailio. I present next a summary of what is planned to happen: 1) Modules from modules_s/ that will be simply moved to modules/ because there is no conflict at all: - avp, avp_db, db_ops, eval, mangler, print, print_lib 2) Modules from modules_s/ that will be renamed and moved modules/ because it is mainly only conflict of name, but not in the backend (e.g., database): - auth_db => auth_dbattrs - because it uses (attribute,value) style to store subscriber profiles, which can be a preferred alternative for many out there - ldap => db2_ldap - it is a DB APIv2 connector to ldap server - bdb => db2_bdb - it is a DB APIv2 connector to berkeley db (to be double-checked) - xlog => xprintf - it uses %spec to print log messages with all the code inside the module - the only reason not discarding it and stick only to modules_k/xlog for the moment is because it is used from other modules, otherwise modules_k/xlog has more features (e.g., syslog facility, color printing, a.s.o) 3) Modules from modules_s/ that will be removed, after absorbing some extra features: - usrloc/registrar => get the ability to store AVPs per contact (modules_k is more advanced in features, like path/gruu/partial outbound support, publish presence states, etc.) - several modules must be reviewed: acc*, radius*, permissions, pike, gflags, options, timer 4) Modules from modules_s/ that will be removed, by moving to obsolete/ directory - the rest of them from modules_s/, like dialog, xcap, cpl-c, presence_b2b, dispatcher, dbtext, domain, etc. The categories and their content are not nailed down, it is the purpose of this discussion to see if something was overlooked. We hope to end this process in max 6 months time - after that will be a unique database schema and a unique set of modules. Flavour system in the Makefile will stay just for letting people choose easier the name of binary, otherwise everything will be the same (e.g., what is compiled, installed, etc). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, Kamailio SIP Server v3.3.2 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.1. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0 or v3.3.1. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.2. For more details about version 3.3.2 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/10/kamailio-v3-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

4 4

sample SCA kamailio.cfg added
by Andrew Mortensen 04 Dec '12

04 Dec '12

I've added a basic working kamailio.cfg with Shared Call Appearances to the SCA docs. It should help clarify where and when it's necessary to have the sca module inspect traffic. Those of you simply wanting to test SCA can just copy and paste the configuration, updating the listen address as necessary. Until the sca module doc on kamailio.org gets updated, you can find the sample configuration in the git master branch, or view it through gitweb at the link below. Best, andrew <http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mo…>

1 0

kamailio vs.ser
by Ovidiu Sas 03 Dec '12

03 Dec '12

Hello all, By inspecting the source code, the only difference that I could see between kamailio and ser flavours is that kamailio has support for the "tm:local-request". Are there any constrains in having the tm:local-request present for ser flavour? Does it make sense to continue to build two flavours? My suggestion would be to focus one one single flavour and go forward with it for the next release. Comments? Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

health check
by Alex Solt 03 Dec '12

03 Dec '12

Hi, Is there a way to perform health check on kamailio application? AS

2 1

LCR Failover
by Daniel W. Graham 03 Dec '12

03 Dec '12

I am having issues with LCR failure route. I have two gateways specified in LCR table, if I change the address of the gateway with highest priority to a bogus IP, the failure route never seems to takes place and call is never routed to second gateway. Also, I have tested successful calls to both gateways by switching the priorities around. Here is what I receive after the first gateway times out: U kamailio:5060 -> asterisk:5060 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK633b44d8;rport=5060. From: "user" <sip:user@asterisk>;tag=as67c0037f. To: <sip:dialednumber@domain>;tag=d7177f97ed67076dddc3163c85ab0656-f2e5. Call-ID: 248d69ed6495d0a65315458e2ba6c528@asterisk:5060. CSeq: 102 INVITE. Server: SIP Proxy. Content-Length: 0. Config: modparam("tm", "fr_timer", 10000) modparam("tm", "fr_inv_timer", 120000) modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") route[LCR] { if (!load_gws(1)) { sl_send_reply("503", "Internal Server Error, Unable to load gateways"); exit; } else { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { sl_send_reply("503", "Service Unavailable, No available gateways"); exit; }; }; exit; } failure_route[LCR_FAIL] { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { t_reply("503", "Service Unavailable, No gateways"); exit; }; exit; }

2 7

Kamailio World - Conference & Exhibition - April 16-17, 2013, Berlin, Germany
by Daniel-Constantin Mierla 03 Dec '12

03 Dec '12

Kamailio project is announcing its first dedicated event - Kamailio World - to take place in Berlin, Germany, during April 16-17, 2013. The event is organized by Asipto (http://www.asipto.com) in collaboration with Fraunhofer Fokus Institute (http://www.fokus.fraunhofer.de). The conference is targeting to present commercial products and services build around Kamailio and related projects, as well as technical workshops of how to use Kamailio for various purposes. Along with the conference will be space for exhibition, that will allow companies to make demos and show cases for participants. For more details, visit the web site of the event: http://conference.kamailio.com Besides Kamailio, you will have the chance to interact with other open source VoIP projects such Asterisk, FreeSwitch, SIP Express Media Server, Homer SIP Capture, Siremis, a.s.o., as well as with commercial vendors in real time communications market. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Jitsi to Jitsi call
by Raj Roy Ghandhi 03 Dec '12

03 Dec '12

Hi Peter, Still I am unable to make call to Jitsi to Jitsi. Call established But no media. But text message works. NAT traversal work with RTPProxy and I can do IPPhone to IPPhone call and voice works very well. Tried with many soft phones and voice did not go through. Did I miss something in the configuration ? Please advice me. Best Regards, Roy.

2 2

Call pickup through a NAT
by Mark Boyce 03 Dec '12

03 Dec '12

Hi All, I've hit my next stumbling block on my mission to get BLF/Call pickup working on Linksys/Cisco phones which are behind a NAT working with Kamailio. I'm using RFC 3265 and pua_dialoginfo to handle subscribes from phones and generate notify packets on status change. The phones are on two different sites behind a couple of NATs. Kamailio is on public IP in the data centre. BLFs work fine. However the <target uri="x.x.x.x."> given in the XML of the notify packet is the that of the Public IP of the NAT so that call pickup fails. I'm guessing that this actually needs to be the address of the Kamailio server? I cannot see where to correct this without breaking tables the server needs to know how to contact the phone when it needs to. Has anyone seen this before? Cheers Mark

2 1

Kamailio SCA Module
by Robert Boisvert 01 Dec '12

01 Dec '12

Andrew, Congratulations on your new position as a Kamailio developer! When I saw your module (SCA) I began testing immediately since it may be a stepping stone to help me get the Polycom barge-in feature working with Kamailio. However, I ran into a few difficulties. Can you assist me? To make your existing code work as a K module I had to change a few lines of code and recompile under modules_k. Specifically, since the K definition of the ucontact structure defines ucontact->aor as str* and the S definition as str**, I changed “&c->aor” to “c->aor” in the following lines of sca_usrloc_cb.c: 73: if ( !sca_uri_is_shared_appearance( sca, c->aor )) { 74: LM_DBG( "%.*s is not a shared appearance line", STR_FMT( c->aor )); 79: &SCA_EVENT_NAME_CALL_INFO, c->aor ) < 0 ) { Since I use PostgreSQL as my SQL engine I created the SCA entries using this code: INSERT INTO version (table_name, table_version) values ('sca_subscriptions', '0'); CREATE TABLE sca_subscriptions ( id SERIAL PRIMARY KEY NOT NULL, subscriber VARCHAR(255) NOT NULL, aor VARCHAR(255) NOT NULL, event INTEGER NOT NULL, expires INTEGER NOT NULL, state INTEGER NOT NULL, app_idx INTEGER NOT NULL, call_id VARCHAR(255) NOT NULL, from_tag VARCHAR(128) NOT NULL, to_tag VARCHAR(128) NOT NULL, notify_cseq INTEGER NOT NULL, subscribe_cseq INTEGER NOT NULL, CONSTRAINT sca_subscriptions_idx UNIQUE (subscriber, call_id, from_tag, to_tag)); I applied the recommendations from your documentation on configuring Kamailio (3.3.1) and inserted the call to “sca_handle_subscribe ()” in kamailio.cfg. Next, I connected two Polycom phones (SP335 and SP650) to my network and configured them to enable presence and use a shared line. Four lines were configured on the phones as shown below: SP335 Line 1 (p11) = private line 6013 SP335 Line 1 (p12) = shared line 1519 SP650 Line 1 (p21) = private line 6014 SP650 Line 2 (p22) = shared line 1519 My tests get inconsistent results on the line LEDs and it seems that they do not work as a shared line. When I ran a tcpdump of the SIP communication (see attached pcap files) I found several anomalies: - Some packets (sca-reboot1 #22, 44, 113, 120 and sca-reboot2 #17, 38) contain spurious NUL characters in the tags section. In all tests it seems to occur in the Expire tag when Kamailio accepts the subscription. - When Kamailio sends call-info NOTIFY the Content-Length tag is missing. - Sometimes the phone does not respond to a call-info NOTIFY event ( sca-reboot1 #23, 45 and sca-reboot2 #18, 39). - I don't have an example here but sometimes the phones respond to the call-info NOTIFY with "481 Call Leg/Transaction Does Not Exist". I think some of my problems are related to these anomalies. Could you help me to determine if this is a problem with the code itself, my failure to properly port it to the K module, or some other issue? Once again, thank you for making this very useful module for Kamailio and for the documentation to make it work. I hope that resolving my problems may help the code to become a part of Kamailio. Sincerely yours, Bob Boisvert

2 1

Linphone presence
by Moacir Ferreira 30 Nov '12

30 Nov '12

I am trying to get Linphone to show presence information. I did my Kamailio setup using just the template from a stardard install and it is working properly with Bria (X-Ten). However, when using Linphone it does not show buddies presence. Any hint? Cheers! Moacir

2 1

Call forking
by Moacir Ferreira 30 Nov '12

30 Nov '12

All the topics/examples I found on the net is not what I am looking for, so here is the question: How do I achieve parallel forking (call all devices) if I have a single user registered using the same account from different devices (i.e.: table phone, iPad and PC)? What I want is to ring all devices a user may be using. Cheers! Moacir

2 1

t_fork_to?
by Juha Heinanen 30 Nov '12

30 Nov '12

core cookbook reads: append_branch Similarly to t_fork_to, it extends destination set by a new entry. The difference is that current URI is taken as new entry. what is t_fork_to? i could not find it in tm README. -- juha

3 3

WITH_PRESENCE charsh Kamailio
by Raj Roy Ghandhi 30 Nov '12

30 Nov '12

Hi, I am running Kamailio 3.4 with web-socket and everything works fine with NAT traversal for IP Phones. But when I enable the PRESENCE it crash with "(3) errors in config file" I have copied the config.cfg to paste-bin at http://pastebin.ca/2257817<http://pastebin.ca/2257815> Please guide me on this issue. Best Regards, Roy.

2 1

Enforced TLS, still some packets are through UDP
by Kamal Palei 30 Nov '12

30 Nov '12

Dear All I ma trying to have all SIP signalling through TLS. Using Kamailio 3.1.5. In a typical setup, endpoints initiates call, messages reach Kamailio proxy, Kamailio proxy forwards these packets to main proxy. In Kamailio.cfg, I have added below code. route { # per request initial checks route(REQINIT); setflag(FLT_NATS); if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.19:443"); exit(); } else { xlog("incoming request\n"); route(WITHINDLG); route(RTPPROXY); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); t_relay_to("tls:115.114.48.19:443"); exit(); } .............. I am finding most of the messages are getting transmitted in TLS. But few ACK and REFER / REFER response packets are still being transported in UDP. Can somebody tell me whats the wrong I am doing here. By any chance route(WITHINDLG); route(RTPPROXY); before t_relay_to("tls:115.114.48.19:443"); can cause this ???? And one more doubt.., 115.114.48.19 I get in route header. How can I extract the value of route header, so that I do not have to hard-code Many thanks in advance. -kamal

3 2

Lua env not initialized in event_route
by Vicente Hernando 30 Nov '12

30 Nov '12

Hello I'm trying to execute a Lua script on kamailio startup. I'm calling the script inside event_route[hable:mod-init], but it fails to be executed with this error: [app_lua_mod.c:167]: Lua env not initialized It happens both with lua_run and lua_dofile. I've played with the order in wich the modules are loaded, but I haven't been able to get rid of that message. Kamailio version is 3.3.0. Do you think this is feasible or should I try a different solution? Thanks in advance Regards, Vicente.

2 1

Presence on Cisco/Linksys handsets
by Mark Boyce 29 Nov '12

29 Nov '12

Hi All, I've been trying to get some Cisco/Linksys SPA phones working with Kamailio (current stable). All seems to be ok apart from getting Presence/BLF/SCA working. The phones are set to Server_Type : RFC3265_4235 Line Key 2 Extension : Disabled Line Key 2 Share Call Appearance : Private Line Key 2 Extended Function : fnc=blf+sd+cp;sub=1001@ourserver.com;ext=1001@ourserver.com Where 1001 is the other phone. The phone appears to be sending event type 'call-info' rather than dialog etc. Phone is being sent '489 Bad Event' and the error in the logs is : presence [subscribe.c:1007]: Unsupported event header field value call-info Whatever I set the Server Type to on the phones it still tries to send call-info events on subscribes Has anyone got Presence working with Cisco SPA handset? The module needed seems to be available on OpenSIPs http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html Thanks! Mark

3 8

Kamailio getting started
by Scott, Matt 29 Nov '12

29 Nov '12

Anybody got a good tutorial or starter project for Kamailio? We have Siremis 3 installed, but there is 0 help on that? Trying to solve 2 scenarios: -asterisk to softphone through kamailio -softphone to asterisk through kamailio. I ultimately want to use Kamailio as an edge router Any help is appreciated. Matt Scott

3 3

Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 29 Nov '12

29 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I’ve setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. ‘kamctl dispatcher show’ shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

2 13

Call from SipML5 -> PSTN
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, >From all of your support now I can call from 1. IP Phone <--> IP Phone 2. Web Page <--> Web Page 3. IP Phone -> PSTN without any issue But when I try to call from Web Page to PSTN then it tries to call sip:00xxxxxx89078@mysipdomain.com and that time out. Trying to figure out how to get this work ? Can anybody guide me on this please. Best Regards, Roy.

6 7

Re: [SR-Users] CRLF from Kamailio server
by Klaus Darilion 28 Nov '12

28 Nov '12

forwarded to mailing-list .... -------- Original Message -------- Subject: Re: [SR-Users] CRLF from Kamailio server Date: Mon, 26 Nov 2012 16:34:26 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Hello, I haven't checked actually the parts for tcp, but recently I added the feature of removing the location records if there is no reply to OPTIONS keepalives in nathelper, and it was only for UDP. Also, I think the nat ping is 4 bytes (zeros) or so. Might be another option from tcp, like: http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_keepalive Cheers, Daniel On 11/26/12 3:54 PM, Klaus Darilion wrote: > IIRC tcp_crlf_ping is only about the Ping-Pong mechanism: client sends > PING=CRLF, server responds with PONG=CRLFCRLF. > > Thus, server should only response on client CRLF. Further the PONG is > always double-CRLF. > > So I suspected it is NAT keepalive - maybe someone added > NAT-keep-alive for TCP :-) > > regards > Klaus > > On 26.11.2012 11:05, Daniel-Constantin Mierla wrote: >> Hello, >> >> this might be actually related to tcp connection keepalive: >> >> http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_crlf_ping >> >> IIRC, nathelper pings only udp contacts. >> >> Cheers, >> Daniel >> >> On 11/26/12 10:54 AM, Klaus Darilion wrote: >>> >>> >>> On 26.11.2012 05:13, Dmytro Bogovych wrote: >>>> Greetings. >>>> May anyone give me advice? >>>> I have working Kamailio with TLS. >>>> It is used by iOS softphone - i developed own one based on >>>> resiprocate. >>>> All signalling is TLS. >>>> Almost everything is fine. >>>> >>>> But looks kamailio sends CRLF messages sometimes. >>>> If it happens when softphone is in background mode - it is killed >>>> by iOS >>>> (as iOS prohibits too often awakes from background mode). >>>> >>>> How I can disable these messages? >>>> I see natping_interval = 30 and ping_nated_only=0 >>>> Should I set natping_interval to zero ? >>> >>> Either set the interval to 0 or set ping_nated_only=1 and make sure to >>> not set the NAT flag for the iphone clients. >>> >>> A problem may be firewalls/NAT between the iPhone and Kamailio, as >>> they may drop the TCP connection is there is no activity for too long >>> time. >>> >>> regards >>> Klaus >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Call from SipML5 -> PSTN
by Peter Dunkley 28 Nov '12

28 Nov '12

You need some sort of media gateway or server. I don't know of any that currently support this. Peter Raj Roy Ghandhi <roy.gandhi(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

memcached loading problem
by Alex Balashov 28 Nov '12

28 Nov '12

Hello, So, I built the memcached module against the latest Git production branch of memcached and libmemcache-1.4.0.rc2 (the latest available), on Debian i386. But when I try to load the Kamailio memcached module: Nov 28 03:47:44 diminuendo-1 kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/memcached.so>: /usr/lib/libmemcache.so.0: undefined symbol: mcm_buf_len I find that very odd, because the symbol is quite clearly exported in the library: diminuendo-1:~# nm /usr/lib/libmemcache.so.0 | grep -i mcm_buf_len U mcm_buf_len Does anyone have any idea what the underlying linking issue is here? -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 4

Websocket with Kamailio 3.2.2 release
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy.

6 17

Eavesdropping SRTP sessions
by Mino Haluz 28 Nov '12

28 Nov '12

Hi, maybe it is not that kamailio related question, but I dont know any other place with such good voip professionals ;) I have kamailio and mediaproxy. Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am forcing clients to use SRTP but it does not support adding any certificate on both sides. SRTP call is working fine. The question is, in this case, is man-in-the-middle attack possible? Maybe I should study SRTP more, but basically, if there are no certificates, there is no method how to be 100% sure that the media goes directly between clients. Is it true? Thanks for response, Mino

2 4

Managing expiration values with memcached
by Alex Balashov 28 Nov '12

28 Nov '12

According to the memcached module docs, setting $mctex(key) can be used to modify the expiration value of a key. Should this be done after or before setting the key itself with $mct(...)? I have tried it before and the assignment didn't work, so clearly the key must first exist in memcache: $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:354]: setting pvar failed Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:407]: assignment failed at pos: (164,37-164,69) OK, so that makes sense. I tried it the other way: $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); But when I look on my memcached server, the expiration value is not being modified. It's still the default 'expire' value set in the memcached modparam. I ran the memcached server with maximum verbosity and verified that no TOUCH command is taking place, and that the SET command for the key contains the default expiration value, not the one I set with $mctex(...). So, it seems to me that the module is not committing this new expiration value when it is written to $mctex(...). Is this a bug, or am I missing something? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 2

problem with kamailio and libradiusclient-ng/freeradius-xs
by Rabary 27 Nov '12

27 Nov '12

Hi all, When I have this line in kamailio.cfg: modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") Kamailio failed to start: kamailio error, failed to start (/var/run/kamailio/kamailio.pid exists). I already try to stop kamailio, an remove this file but don't work In log,I have: Nov 27 20:13:39 voip-core-1 kernel: [14318.988658] kamailio[8370]: segfault at 0 ip 00007f0b0b94ca5d sp 00007fffa08f9460 error 4 in libradiusclient-ng.so.2.0.0[7f0b0b946000+c000] Nov 27 20:13:39 voip-core-1 kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe When I remove just this line and restart kamailio, it works, I don't know what is the problem because before I can do accounting with freeradius with the same config I have kamailio 3.3.2, kamailio-radius-modules 3.3.2,libradiusclient-ng2 0.5.6-1.1 and on 2nd server freeradius-xs and freeradius-xs-mysql 1.1.7-4 I put in attached file a few config of kamailio and radiusclient Regards -- Rabary Teddy Inutile d'imprimer ce mail

1 0

Disconnection from Kamailio
by Dmytro Bogovych 27 Nov '12

27 Nov '12

Greetings. I observe strange behavior with Kamailio. My iphone 3gs run softphone based on resiprocate. First 10-20 minutes everything works fine. After it the connection is closed from server (select() loop returns read notification, read attempt returns zero bytes). Ok, softphone closes old socket and create new one by new registration. But this connection is terminated from server in 4-5 seconds after registration finish (200 OK). Softphone makes next attempt to register - connection is closed after registration again. So it is infinite loop. After several attempts iOS kills application as it awakes too often in background mode. I have not logs from server yet; but maybe anyone had similar issue? Thank you :)

1 0

send 100 provisioning delayed
by pars3c 27 Nov '12

27 Nov '12

Hi , The proxy send the "automatic" provisioning message 100 "Trying" , near 1-2 seconds later the INVITE message, so the client send 3 times the same invite (correct , 500 ms timer). Is there any parameter to set , for the time between the INVITE and "100" message ? Is correct to disable the automatic send, and set after the invite, sl_send_reply("100", "Trying"); ? Thanks

2 1

Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 27 Nov '12

27 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I've setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. 'kamctl dispatcher show' shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

1 0

Kamailio crash
by Ricardo Martinez 27 Nov '12

27 Nov '12

Hello In the last three days my kamailio process has crashed two times. I obtained the core file and see it, this is the output. Can someone check what is the problem? Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 56 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 #1 process_mi_params (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1057 #2 mi_terminate_dlgs (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1165 #3 0x00007fc789996370 in run_mi_cmd (rx_sock=11, tx_sock=12) at ../../lib/kmi/mi.h:77 #4 mi_datagram_server (rx_sock=11, tx_sock=12) at datagram_fnc.c:525 #5 0x00007fc789998509 in datagram_process (rank=<value optimized out>) at mi_datagram.c:346 #6 0x00007fc78999a09c in mi_child_init (rank=<value optimized out>) at mi_datagram.c:296 #7 0x00000000004dbd61 in ?? () #8 0x00007fc78d99aab8 in ?? () #9 0x00007fc78d0181cd in pkg_proc_update_real_used (data=<value optimized out>) at pkg_stats.c:141 #10 0x000000000052b423 in ?? () #11 0x00007fc78d8a2830 in ?? () #12 0x0000000000000000 in ?? () (gdb) I’m using kamailio [root@pxh ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 2a00c6-dirty with dialog module, hash tables, radius authorization. Hope someone could help me. Regards, Ricardo.-

2 8

plans for next major release
by Daniel-Constantin Mierla 27 Nov '12

27 Nov '12

Hello, following the devel IRC meeting from last week, I am starting the discussion about the next major release planning. There were two relevant options: 1) freeze by end of January (2013) and release by March 2) freeze by end of December (2012) and release by February Many devels announced quite some work ahead, so the 1st option seemed not to be enough for finishing intended work, which may add some extra delays (several estimated to finish by end of February or March). So the 2nd came into debate, to do an early release as many people seemed very interested in getting websockets support out in a stable release. Both of the options fit in our releasing policy (8 to 10 months), first one with potential to delays. We do have now lot of new stuff in devel branch, with other ongoing work planned to be pushed before end of year. A suggestion was to do the following major release a bit shorter, if it proves to collect a lot of additions quickly -- maybe by summer 2013. As not all devels were in the meeting, but also because it impacts the user community, I am looking to get some feedback on the mailing lists as well. That will allow to make the decision quicker based on showed interest. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

One active registration per sip instance.
by Krishna Kurapati 26 Nov '12

26 Nov '12

Hi, I have 3.x Kamailio with GRUU support. I see SIP Instance in the contact. However, I see a problem. When I kill the SIP client on a device and restart, it sends registration on a different port. The server instead of updating the contact in user location table, is creating a new contact for the same instance. Do I need to turn on something to get the expected behavior? Here is the output from kamctl ul show. Thanks Krish Kura AOR:: 100227074 Contact:: sip:100227074@10.132.60.70:50845;transport=TCP;ob Q= Expires:: 3132 Callid:: ATZJcr.HqXxwWdsPmanjERetcdeRVeZy Cseq:: 4399 User-agent:: n/a Received:: sip:166.137.120.37:48873;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-50e-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50839;transport=TCP;ob Q= Expires:: 3029 Callid:: oYsIYUyP3N77MKSoQ3EDChEph.lD4fjB Cseq:: 37489 User-agent:: n/a Received:: sip:166.137.120.37:61763;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-507-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50803;transport=TCP;ob Q= Expires:: 1527 Callid:: b4IeJXybn6Yr.jB8sc-Kz8CJ0kt0hNbO Cseq:: 30014 User-agent:: n/a Received:: sip:166.137.120.37:36879;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5c-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50818;transport=TCP;ob Q= Expires:: 2199 Callid:: DM0Z1d-ZenGHknjmT.7Cle5cos7Uhs7H Cseq:: 22942 User-agent:: n/a Received:: sip:166.137.120.37:39471;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f6f-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50815;transport=TCP;ob Q= Expires:: 1833 Callid:: VJy3SNE66U1U971u5BEeqbYJi6a-puT- Cseq:: 64347 User-agent:: n/a Received:: sip:166.137.120.37:35872;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5a-2 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1

2 4

new developer: Andrew Mortensen
by Daniel-Constantin Mierla 26 Nov '12

26 Nov '12

Hello, I want to announce that a new developer got GIT write access to repository: Andrew Mortensen from University of Pennsylvania, USA - he has contributed a new module: sca (Shared Call Appearances for Broadsoft extensions). The module is right now in a personal branch, soon to me merged in master branch. His git commit id is: admorten My warm welcome and looking forward to future work within the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

ul.dump Problem
by Spencer Thomason 26 Nov '12

26 Nov '12

Hello, I'm attempting to dump the contents of the usrloc cache and I'm receiving the following error: sercmd> ul.dump error: 500 - Internal error creating aor struct sercmd> mi ul_dump error: 500 - Internal server error processing 's': buffer too small (overflow) (-2) It seems to work fine when there are only a small number of registrations (about 20 or so) but then fails when the number increases. It there a setting I need to increase somewhere? Thanks, Spencer

2 4

problem with $time Pseudo-Variables on 3.3.2
by Uri Shacked 26 Nov '12

26 Nov '12

Hi, I upgraded to 3.3.2 On 3.2 i used the variable $time() to get the current time for some process. Every 2 minutes a RTIMER process started, checked the $time(hour) and $time(min) and on a certain time executed something. This process stopped working on 3.3.2 When i debug, i see that the $time() variable keeps on giving me the same hour and minute that was given on the first run. Meaning the $time() variable does not give the current time. I notice also that the $Ts that on 3.2 gave a current time stopped doing it and $TS does it now.... Any help here? Is it OK and i need to check time another way? Or something in the $time() has changed...? Thanks, Uri

2 6

CRLF from Kamailio server
by Dmytro Bogovych 26 Nov '12

26 Nov '12

Greetings. May anyone give me advice? I have working Kamailio with TLS. It is used by iOS softphone - i developed own one based on resiprocate. All signalling is TLS. Almost everything is fine. But looks kamailio sends CRLF messages sometimes. If it happens when softphone is in background mode - it is killed by iOS (as iOS prohibits too often awakes from background mode). How I can disable these messages? I see natping_interval = 30 and ping_nated_only=0 Should I set natping_interval to zero ? Thank you :)

3 4

Re: [SR-Users] Forwarding Relaying of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi All Alex responded to my question saying the following: "You can relay SIP messages of any type just the same way as you do with INVITEs -- e.g. t_relay(), forward(). But, proxying registration messages can have complications." can anyone give me some pointers or a list of some of these complications so that I can further research to see how its is done. I will attempt using the t_relay to view the issues, just need some idea more specifically what to look for! Thanks

2 1

Re: [SR-Users] Re-Directing of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi Can anyone point to a good example of Kamailio performing the function of Re-direct server. We would like to forward messages onto another domain for authentication.

4 3

Difficulties with with loose_route() when trying to handle non-compliant ACK
by Richard Brady 24 Nov '12

24 Nov '12

Hi guys I have a multihomed Kamailio proxy sitting between two B2BUAs on separate networks and record-routing all dialogs. The problem I have is that when one of these devices receives a 200 OK, it does not populate the RURI of the ACK correctly. Instead of taking it from the Contact header on the 200 OK, it uses the user part from the Contact header and sets the domain to the proxy IP. It then also populates the Route headers. The ACKs are below and the IPs are: 10.152.1.92:5060: UAC on outside 10.200.70.100:5060: proxy outside interface 192.168.242.100: proxy inside interface 192.168.242.102: UAS on inside This the ACK message going into and coming out of the proxy. # U 10.152.1.92:5060 -> 10.200.70.100:5060 ACK sip:natted_ua*9197**192.168.242.102*5080*udp@10.200.70.100 SIP/2.0. Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK-2280-1-5. Route: <sip:10.200.70.100;r2=on;lr=on;nat=yes>,<sip:192.168.242.100;r2=on;lr=on;nat=yes>. From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: sip:sipp@127.0.1.1:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.200.70.100:5060 -> 192.168.242.100:5060 ACK sip:192.168.242.100;r2=on;lr=on;nat=yes SIP/2.0. Via: SIP/2.0/UDP 10.200.70.100;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=10.152.1.92;branch=z9hG4bK-2280-1-5. Route: . From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: <sip:sipp@127.0.1.1:5060;alias=10.152.1.92~5060~1>. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. So I am trying to understand why it is trying to relay to itself (I have mhomed=1) and why it is rewriting the RURI as if it is a strict router. Currently decode_contact() is disabled but enabling doesn't seem to help. Unless there is a very specific place where it belongs. Any advice would be hugely appreciated. I can always paste logs / configs / traces. Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

2 3

can not make calls with kamailio 3.3 and asterisk-10.7.0
by James zhu 24 Nov '12

24 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com

1 0

Using uac_req_send function
by Freddy C 22 Nov '12

22 Nov '12

Hello, I'm using uac_req_send () to send a request generated in Kamailio (eg Example 1.21. UAC module) $ uac_req (method) = "OPTIONS"; $ uac_req (ruri) = "sip: kamailio.org"; $ uac_req (furi) = "sip: kamailio.org"; $ uac_req (turi) = "sip: kamailio.org"; uac_req_send (); However, I need to capture the responses of this request with failure_route TM module. Is there a way to capture these responses when I'm using uac_req_send() function? I appreciate your comments. Freddy

2 1

Next IRC devel meeting
by Daniel-Constantin Mierla 22 Nov '12

22 Nov '12

Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 10

How does SIP transfer work?
by Grant Bagdasarian 22 Nov '12

22 Nov '12

Hello, I've been searching the internet to find an explanation on how SIP transfer works using Re-INVITE and/or UPDATE, but I can't seem to find a good source. >From what I understand(and this is the way we do it), the following happens: Bob=Caller Alice=Called John=Transfer party 1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK. 2) Alice transfers the call to John using Re-INVITE. a. Alice calls John. The usual INVITE,Trying,200 OK, ACK. b. Alice Re-INVITEs Bob using INVITE with adjusted SDP. 3) Bob is connected to John through Alice in some magical way. I'm guessing because the SDP has been changed and for some reason the RTP stream flows between Bob and John through Alice? Is this correct? If not, perhaps someone could explain it to me from scratch. Maybe useful to know that we are using Cisco equipment for call handling (VXML and TCL scripts). Thanks, Grant

3 2

Forward Register Message
by kellon＠sis-grenada.com 21 Nov '12

21 Nov '12

Hi All Can anyone point to documentation tutorial on howto forward statefully a REGISTER message to another SIP Server. I have looked at http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, However I'm getting errors with the uac_req commands even though I have loaded the uac.so module. .

2 1

Shared Call Appearances module
by Andrew Mortensen 21 Nov '12

21 Nov '12

I've been working on a Shared Call Appearances module for the past several months. It implements the Broadsoft SCA feature as laid out in Broadworks SIP Access Side Extensions Interface Specifications document. (Another implementation of the same feature was added to Freeswitch a few years ago, but we didn't want to use Freeswitch.) We've been testing and improving the module over the last month, and I think it's ready to share. The module is available at GitHub here: <https://github.com/fitterhappier/sca> To date, we've only tested the module with Polycom handsets. I'm happy to answer questions. I hope this will prove useful for others. Best, andrew

6 18

cost script
by Ruben Mejia 21 Nov '12

21 Nov '12

IPTEL We needhelp or cost for the development a script for kamailio which can do the following: Register kamailio to another gateway using UAC modules. route all traffic coming towards a private ip 10.x.x.x We will use a postgres database We have installed kamailio 3.1 with postgres support and UAC modules Rubén Mejía Telefonia Cabletamps

1 0

I can not make call from kamailio to asterisk by 3.3
by James zhu 21 Nov '12

21 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com > From: sr-users-request(a)lists.sip-router.org > Subject: sr-users Digest, Vol 90, Issue 72 > To: sr-users(a)lists.sip-router.org > Date: Wed, 21 Nov 2012 08:58:40 +0100 > > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Shared Call Appearances module (Daniel-Constantin Mierla) > 2. Re: Shared Call Appearances module (Andrew Mortensen) > 3. Re: LCR weight (Juha Heinanen) > 4. problem with $time Pseudo-Variables on 3.3.2 (Uri Shacked) > 5. Re: Dialog module not decrementing profile size > (Grant Bagdasarian) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 20 Nov 2012 22:56:10 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <50ABFC7A.2070602(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 11/20/12 10:41 PM, Ovidiu Sas wrote: > > On Tue, Nov 20, 2012 at 4:22 PM, Andrew Mortensen > > <admorten(a)isc.upenn.edu> wrote: > >> On Nov 20, 2012, at 3:16 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >> > >>> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >>>> > >>>>> Hello, > >>>>> > >>>>> thank for contribution! > >>>>> > >>>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> wrote: > >>>>>> > >>>>>>> Also quick read at the readme, it looks like it does not support multi-domain setups? > >>>>>> Not yet, no. I don't think it would take much work, though. > >>>>> I see it binds to usrloc, what information needs from there? We have two such modules right now, it would be good to make it work with both, kamailio's one has more features in regard to GRUU and partial outbound extensions. > >>>> It's registering for contact expiration and deletion events. The callback terminates subscriptions for the expired/deleted contact. This should probably be configurable. > >>> Ok, so it is about when a contact record in removed from location. > >>> > >>> Just to avoid misunderstandings, do you consider to make configurable this termination of subscriptions based on location? > >> Yes. I can imagine a setup where the location data wouldn't be saved on the host handling SCA. > >> > >>> Also, you need only the AoR and contact address in a callback for location record removal event (un-registration or registration expiration), right? > >> Correct, although Ovidiu makes a good point about (well-behaved) clients unsubscribing at the time of unregistration. In registering for usrloc callbacks, I'm trying to handle a couple problems. First, I'd like to avoid unnecessarily sending and retransmitting NOTIFYs to offline clients. More importantly, I want to be able to hear about expired registrations so I can clear appearance state on the rest of the subscribers. > > For clearing appearance state, you should rely on the dialog module > > and register a callback on the dialog module. When a dialog expires, > > you can clear up the stale appearances. > > Also, by enabling the sst module, you can have a tight control over > > the duration of the dialog. > > This will make the implementation cleaner and more robust. > > > I will not jump to such conclusions. > > While dialog module is designed to track active calls, does not mean > everything has to be on top of it. Just for example, dispatcher module > has an embedded lightweight call tracking system. > > No idea here what the broadsoft extensions require, afaik, there were > some custom specs. It is no problem to have a dedicated dialog tracking > system that is better designed for a particular functionality. We do > have other duplicated systems, for example least cost routing, nat > traversal, a.s.o. > > Also, optimizations and reuse of other parts can be done later if found > to be better ways. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > > > ------------------------------ > > Message: 2 > Date: Tue, 20 Nov 2012 17:17:38 -0500 > From: Andrew Mortensen <admorten(a)isc.upenn.edu> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <65EAE745-8966-4848-870E-6A50246E0AB7(a)isc.upenn.edu> > Content-Type: text/plain; charset=iso-8859-1 > > > On Nov 20, 2012, at 3:43 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: > > > Hello Andrew, > > > > First of all, thank you for sharing your work. > > I was following this thread and I have a couple of questions. Why do > > you need to bind to the usrloc module? The subscription itself should > > be sufficient because if a phone will unregister, it will also > > unsubscribe, > > Yes, that's good point. However, if the phone goes off-line instead of unregistering, the contact's registration will expire, and no unsubscribe will take place. As I mentioned in my reply to Daniel, I'm most concerned about catching this so I can release any appearances seized by the expired/deleted contact, and notify other members of the group. I'm certainly open to alternatives. > > > which leads me to the following question: why not add a > > new dedicated module for call-info presence and reuse the existing > > infrastructure from kamailio for handling subscriptions/presence. In > > this case, your module should just push PUBLISH events to the presence > > server and the server will automatically send out notifications for > > subscribers that subscribed to call-info events. > > We've worked extensively with the existing presence & pua modules, albeit primarily dialog;sla using OpenSIPS behind the proxy. (Thanks, Anca!) > > SCA's unusual entanglement with call processing made me hesitate about building on top of the existing presence module. For a first pass, I also felt working with an entirely distinct module gave me more control and transparency during development of a very loosely documented event package, especially as I became more familiar with the available API. > > I don't have any objection to revisiting design decisions, of course. I'm sure the module will continue to evolve, and it would be nice to eliminate redundancy if possible. > > > Based on your README file, you inspect SIP requests/replies and based > > on the presence of the call-info header, you create call-info events. > > This is great for sharing the appearances between phones, but how do > > you perform the retrieval of a call that was put on hold? Are you > > using a dedicated B2BUA behind kamailio? > > No B2BUAs are involved. > > The INVITE retrieving a call held by another member of an SCA group has a particular set of characteristics: RURI, To and From URIs are all the SCA AoR; new unconfirmed dialog (no to-tag); and a Call-Info header referring to the index of the held call. The module detects this type of INVITE, looks up the dialog associated with the information in the Call-Info header, and injects a Replaces header with the dialog of the held call before relaying it to the remote party. The remote party must support RFC3891. I've only worked with Polycoms and Cisco gateways to this point, and both do support that RFC. > > I'm very interested in hearing from users with Cisco (and Snom and Aastra?) SCA setups. I have no doubt they'll find bugs resulting from assumptions I've made in the code because I've only tested with Polycom, not least the dependency on RFC3891 support. > > Thanks for your feedback! > > Best, > andrew > > > > > On Tue, Nov 20, 2012 at 3:16 PM, Daniel-Constantin Mierla > > <miconda(a)gmail.com> wrote: > >> > >> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>> > >>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> > >>> wrote: > >>> > >>>> Hello, > >>>> > >>>> thank for contribution! > >>>> > >>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>> > >>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> > >>>>> wrote: > >>>>> > >>>>>> Also quick read at the readme, it looks like it does not support > >>>>>> multi-domain setups? > >>>>> > >>>>> Not yet, no. I don't think it would take much work, though. > >>>> > >>>> I see it binds to usrloc, what information needs from there? We have two > >>>> such modules right now, it would be good to make it work with both, > >>>> kamailio's one has more features in regard to GRUU and partial outbound > >>>> extensions. > >>> > >>> It's registering for contact expiration and deletion events. The callback > >>> terminates subscriptions for the expired/deleted contact. This should > >>> probably be configurable. > >> > >> > >> Ok, so it is about when a contact record in removed from location. > >> > >> Just to avoid misunderstandings, do you consider to make configurable this > >> termination of subscriptions based on location? > >> > >> Also, you need only the AoR and contact address in a callback for location > >> record removal event (un-registration or registration expiration), right? > >> > >> > >>> > >>>> Would you consider adding it to main git repository and maintaining it > >>>> there? > >>> > >>> Yes, that would be great. > >> > >> > >> We will prepare write access for you. > >> > >> > >>> Would you prefer to have it in modules? I started work on it in > >>> modules_s, but there's no particular reason it has to stay there. > >> > >> To make it work with both variants of usrloc modules, I think it requires > >> some split of the code. IIRC, both modules still use same names for > >> structures, so including files from both of them will fail. > >> > >> One solutions I am thinking of is to add two small modules, one that binds > >> to usloc(k) and the other to usrloc(s). Each of these two modules will > >> export a function that allow registering callbacks for execution when a > >> location record expires, giving the aor, contact and event type. The > >> existing module will use the intermediate module instead of directly binding > >> to usrloc structures. > >> > >> Usrloc module from kamailio has more enhancements in regards to standard > >> extensions for location services, still it lacks the feature of being able > >> to store variables per contact record. It is the reason that kept back the > >> removal (upon merging) of usrloc module from ser. The solution with > >> intermediary modules should work until we have a merged usrloc module. > >> > >> What do you think about this proposal? > >> > >> Cheers, > >> Daniel > >> > >> > >> -- > >> Daniel-Constantin Mierla - http://www.asipto.com > >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > >> > > > > > ------------------------------ > > Message: 3 > Date: Wed, 21 Nov 2012 01:04:52 +0200 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: Re: [SR-Users] LCR weight > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <20652.3220.931000.738348(a)sip.test.fi> > Content-Type: text/plain; charset=us-ascii > > Camila Troncoso writes: > > > scripts are provided in lcr/utils directory that can be used to check > > the probabilities resulting from a given set of weight values. Same can > > be done with command 'kamctl eval_weights'. > > > > I really don't understand what is the probability finally assign to each > > gateway when they have the same priority. > > All I can see is that it doesn't correspond to percentage of capacity from > > the example above , but how can I determine what value gives the percent > > that I want? > > > > Please give an example like: > > > > Rule_id Gw_id Priority weight > > 36 68 1 60 > > 36 69 1 14 > > 36 70 1 13 > > 36 71 1 13 > > you have to figure that out yourself using the script referenced in > readme. > > -- juha > > > > ------------------------------ > > Message: 4 > Date: Wed, 21 Nov 2012 09:55:17 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] problem with $time Pseudo-Variables on 3.3.2 > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhQn1r_ke72-u2is8MyRXUV6KkLnZqYwxL5eZG44yYG_2Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I upgraded to 3.3.2 > > On 3.2 i used the variable $time() to get the current time for some process. > > Every 2 minutes a RTIMER process started, checked the $time(hour) and > $time(min) and on a certain time executed something. > > This process stopped working on 3.3.2 > > When i debug, i see that the $time() variable keeps on giving me the same > hour and minute that was given on the first run. > > Meaning the $time() variable does not give the current time. > > I notice also that the $Ts that on 3.2 gave a current time stopped doing it > and $TS does it now.... > > Any help here? > > Is it OK and i need to check time another way? Or something in the $time() > has changed...? > > Thanks, > > Uri >

1 0

dialog ka_timer and ka_interval problem
by Kristofer Signer 21 Nov '12

21 Nov '12

Hello all, first of all I would like to thank you all your help. I've tried to implement dialog keep-alive messages but cant make kamailio send more then just one SIP OPTION request to each end. The OPTION request is sent after ka_interval time but no more OPTION requests are sent during the call. How should this be implemented? and was is the difference and relation between ka_timer and ka_interval. tanks ./Kristofer

2 2

LCR weight
by Camila Troncoso 20 Nov '12

20 Nov '12

Hi, I have a question regarding LCR module and their weight variable. If I have 4 gateways for the same prefix, and with the same priority: Rule_id Gw_id Priority weight 36 68 1 60 36 69 1 14 36 70 1 13 36 71 1 13 The weight variable acts as a percentage of utility of that gateway? If for example I have 100 calls, the gateway with gw_id 68 will receive 60 calls , and the other 14, 13 and 13? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 3

Getting started with Kamailio
by Joel Smith | VOZELIA 20 Nov '12

20 Nov '12

Hi, First of all say that I'm new to Kamailio. I've been working with asterisk for a while now and it's time to get started with the sip router project. My idea for a start is to reduce the load of our asterisk servers and once we have good knowledge on how kamailio works, migrate some of our services. I've setup a server in our lab with kamailio 3.3.0 and siremis 3.3.0, created a user (subscriber) and registered with a softphone. So far OK. Now I'm trying to make a call and route it through one of our asterisk servers for testing. To do this I need LCR module (thanks Sébastien, I wasn't subscribed to this list yet) and configure it. After reading http://kamailio.org/docs/modules/stable/modules/lcr.html#id2553110 I've added in kamailio.cfg: #!define WITH_LCR #!ifdef WITH_LCR loadmodule "lcr.so" #!endif #!ifdef WITH_LCR # ----- lcr params ----- modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") #!endif And now I see LCR module being loaded at startup. But, I don't understand what gw_uri_avp, ruri_user_avp or tag_avp mean. Where can I find some doc/manual to understand what AVP means and does? (For example the $avp(i:709), etc.) I've added a LCR GW, LCR TARGET and LCR RULE with the following data (I have no idea if it's completely wrong): mysql> select * from lcr_gw; +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 0 | TEST | XX.XXX.XXX.X | test12345678.testing.com | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 0 | NULL | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule; +----+--------+--------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+----------+-------------+---------+---------+ | 1 | 1 | .* | | | 0 | 1 | +----+--------+--------+----------+-------------+---------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 1 | +----+--------+---------+-------+----------+--------+ 1 row in set (0.00 sec) mysql> Am I still missing basic stuff to be able to place a call ? (To get started I'd like to route all calls to our Asterisk server) I'm glad to read documentation so If all of this is already somewhere please guide me cause I haven't found too much... Thanks in advanced. Regards, Joel.

4 4

t_branch_timeout()
by Alex Balashov 20 Nov '12

20 Nov '12

I am using the latest pull of branch 3.3, and t_branch_timeout() doesn't work if the previous branch resulted in a reply (i.e. a negative final reply). That is to say, t_branch_timeout() == FALSE on a genuine timeout, as long as the preceding branch provided a result (such as 404). t_branch_timeout() == TRUE if this is a timeout, on the first branch. In other words, it appears that if t_any_replied() == TRUE, then t_branch_timeout() == FALSE, even for this branch, and even if the branch really did time out (no response at all from destination). That's not how the documentation says it should behave. Am I doing something wrong? If so, how do I find out if _this particular_ branch timed out, from the failure_route[]? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 3

temporary SIP user
by Bolang 20 Nov '12

20 Nov '12

Hi all, is there any standard mechanism to create a temporary user? i know i can create it and then delete it after some amount of time. but, i'm looking for an established/standard way of doing this. Ideally, the user will be automatically deleted after unregistered. thanks

2 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 20 Nov '12

20 Nov '12

I'd be curious to know if your performance might actually be higher, owing to improved CPUs since the study was done, hyperthreading, more frames per NIC interrupt, etc. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Thank you, setting ulimits worked! And the performance is the same as stated in the document I mentioned! :) One more thing, I found these errors in syslog: Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:211]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5270]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location This is possibly something related to my sipp scenario, this is the INVITE sent from sipp <![CDATA[ INVITE sip:800@perftest.vm SIP/2.0 Via: SIP/2.0/[transport] 10.0.2.36:[local_port];branch=[branch] From: "700" <sip:500@10.0.2.36>;tag=[call_number] To: <sip:800@perftest.vm> Call-ID: [call_id] CSeq: 1 INVITE Contact: "700" <sip:700@10.0.2.36:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.2.36 s=- c=IN IP[local_ip_type] 10.0.2.36 t=0 0 m=audio [auto_media_port] RTP/AVP 8 a=rtpmap:8 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ]]> Do you see something that could cause this error ? Otherwise the call is initiated ok, but I really dont understand what is so strange to kamailio in this INVITE. On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >> You mean on the proxy side? I'm running rtpproxy as root, limits are >> still applied ? ulimit -s unlimited should do the trick ? > > Yes, they usually applied even for superuser, and yes - this should > help (if that's the issue). > > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

6 14

Clarification on is_known_dlg()
by David 20 Nov '12

20 Nov '12

Hello, I am running Kamailio 3.2.3. I am storing dialogs in the database table dialog. loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) later I have : if ( is_known_dlg() ) { } I would like a clarification on how the matching is done. Given that I have realtime table setup, does that mean that dialogs that are in the table will return true when I call is_known_dlg() ? Does that mean that if the dialog is NOT in the table, is_known_dlg will return false ? Thanks for the clarification ! David

2 1

UAC and realm
by Johan Wilfer 20 Nov '12

20 Nov '12

Hi, I've done some tests with the UAC module to authenticate to a remote proxy. I've based my config on this example: http://docs.huihoo.com/opensips/tutorials/uac/ar01s06.html (example 9) I have found that if I send a call from a asterisk via kamailio to my remote proxy the realm the remote proxy uses is the ip-address of the asterisk-server (aaa.bbb.ccc.ddd). Like this: WWW-Authenticate: Digest nonce="1353399363:8d2317487ee1521328d0e2237e444e2d",algorithm=MD5,realm="aaa.bbb.ccc.ddd",qop="auth",stale=false. If I have a param like this, the auth works: modparam("uac","credential","username:aaa.bbb.ccc.ddd:password") But as I would like to use more than one remote proxy (=more than one provider) per asterisk server this needs some more work. Is there a way to get UAC to ignore the realm from the remote proxy and send the auth anyway? Alternate paths I've examined: 1. Rewrite from-domain to get the provider to use another realm: modparam("uac","credential","username:example.com:password") uac_replace_from("sip:$fU@example.com"); Works, but I don't think this is a very good solution. 2. Use auth_username_avp / auth_password_avp / auth_realm_avp to feed the uac_auth()-function the right credentials. modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") in the failure route: $avp(s:uac_user) = "username"; $avp(s:uac_pass) = "password"; $avp(s:uac_realm) = "aaa.bbb.ccc.ddd"; This works as well, but I would like to either get the realm the remote proxy sends and use it or make uac ignore the realm. The realm is unimportant as we already know what proxy we are talking to. I've tried $hdr(WWW-Authenticate) but it evaluates to null. Any ideas how to do this? Thanks! -- Johan Wilfer

2 2

Re: [SR-Users] Returning problem
by Moacir Ferreira 19 Nov '12

19 Nov '12

I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com To: sr-users(a)lists.sip-router.org Date: Mon, 12 Nov 2012 14:42:31 -0200 Subject: Re: [SR-Users] Returning problem This is more a system initialization problem, so, take a look in your /etc/rc<run-level>.d directory (or the correspondent one), specially in the S* files and find out the start sequence. There You'll find the answer. Edson. Em 09/11/2012 17:03, Moacir Ferreira escreveu: > My kamailio start script has the following BEGIN INIT INFO: > > # > ### BEGIN INIT INFO > # Provides: kamailio > # Required-Start: $syslog $network $local_fs $time mysql > # Required-Stop: $syslog $network $local_fs mysql > # Default-Start: 2 3 4 5 > # Default-Stop: 0 1 6 > # Should-Start: postgresql mysql radius > # Should-Stop: postgresql mysql radius > # Short-Description: Start the Kamailio SIP proxy server > # Description: Start the Kamailio SIP proxy server > ### END INIT INFO > > However, when I boot my server Kamailio will start before mysql. So, > after boot I can start Kamalio but at boot time it will fail. > > Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. > All the code has been compiled at the server. > > Cheers! > > Moacir > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 5

how to use uac_auth in module UAC
by Freddy C 19 Nov '12

19 Nov '12

Hello, I have a question about UAC and function module "uac_auth" for a REGISTER request. My idea is to use this when I get a 401 unauthorized response. This answer is captured from failure_route. Then generate the header Authorized and forward the request REGISTER to Asterisk. I could help with some examples?. Freddy Condori Ch Ingeniero Investigador Telefonía IP

1 0

Sync nonce between various servers
by Andreas Granig 19 Nov '12

19 Nov '12

Hi, There are lots of parameters controlling the creation of nonce values on a server, and I'm curious if there is a way to kind of "sync" them between servers. The use case would be to have a UA send for example its registration to Proxy1. Proxy1 would challenge it, UA will send the registration again, this time with credentials. Proxy1 would look up the user based on $au/$ar in the subscriber table, and if it's not found, will look up the responsible proxy from another table (with key being $au@$ar), forward it to Proxy2, which then would be able authenticate the user. The reason for this is that the auth credentials are unique across all servers and reliably identify a user, whereas for example From could be something else (e.g. in case of an IP-PBX sending a CLI in the From-userpart). Challenging the user on the second proxy again would theoretically be possible, but if the UA gets a 401 twice (once from Proxy1, once from Proxy2), it'll most likely pop up a password form for soft-clients, so I want to avoid that. Any ideas how to accomplish that? Andreas

4 9

Problem routing to voicemail
by Christophe ROY 19 Nov '12

19 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. What could be the problem? Thanks for your help Christophe Below is my kamailio.cfg: #!WITH_DEBUG #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - 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/dokuwiki/doku.php # 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 # # *** 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 # ###!define WITH_NAT # *** 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 ###!define WITH_ASTERISK ###!define WITH_VOICEMAIL #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL ####### 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. #!define DBURL "mysql://openserrw:openserrw@192.168.14.28/openser" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asteriskpwd@192.168.14.28/asterisk" #!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 ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL6 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="mydomain.corp" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:192.168.14.24:5060 #listen=tcp:192.168.14.25:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060 #!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" #!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 = "192.168.14.32" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.14.32" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.14.24" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.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" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!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" #!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 #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- 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) # suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967) # ----- 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) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- 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", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- 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 #chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- 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 # ----- speedial 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) # use caching modparam("domain", "db_mode", 1) # 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:192.168.14.25:22222") # ----- 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@teopad-toip.corp") # 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", "/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", 1) #!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); xlog("L_INFO","apres REQINIT"); # NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); xlog("L_INFO","apres t_check_trans"); # authentication route(AUTH); xlog("L_INFO","apres 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(); xlog("L_INFO","apres RECORD ROUTE"); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE"); # dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT"); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE"); # handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN"); # user location service route(LOCATION); xlog("L_INFO","apres LOCATION"); route(RELAY); xlog("L_INFO","apres RELAY"); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("L_INFO","reply error"); 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 $fu (IP:$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] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else 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 xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); 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(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar"); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # 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 #edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!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); } } # 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 #edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif 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; # } { #edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif { # ldap search if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?teopad-Sip-Username,teopadSipPassword?one?(teopad-Sip-Username=$fU)")) # if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?sAMAccountName,?one?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (uid=$fU)"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("teopad-Sip-Username/$avp(s:username)"); ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); 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")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ; if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } } #consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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 # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); 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")) { route(TOVOICEMAIL); exit; } #!endif } #edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 3

Problem with include_file quotes
by Pirjo Ahvenainen 19 Nov '12

19 Nov '12

Hi All, I've been away from Kamailio for a few months, but recently I started working with it again. I could use a hand in a small problem: I'm using include_file statement to give some structure to my config file, but when checking the config I keep getting error message: 0(4290) : <core> [cfg.lex:1554]: included file name must be between quotes: "defined_modules.cfg" I'm using Kamailio 3.3.2 (version: kamailio 3.3.2 (x86_64/linux) 78d4b1) on Ubuntu. And here's the beginnign of the config file: #!KAMAILIO # include_file "defined_modules.cfg" include_file "defined_values.cfg" include_file "global_parameters.cfg" Those look like quotes to me, but could the problem be something like encoding of the config file itself, something in the included files, or is there something else I can check? I don't have extra whitespaces etc, perhaps a typo I just don't see? Cheers, Pirjo

2 3

How to get dialog variable externally?
by Mino Haluz 19 Nov '12

19 Nov '12

Hi, I'm using dialog module and I set some dlg_var for every dialog. Is it possible to get this variable value with MI command somehow ? I know to get list of dialogs using dlg_list only. Thanks! Mino

2 1

db_sqlite bugs
by Pedro Antonio Vico Solano 19 Nov '12

19 Nov '12

5 11

Planning for memory
by Moacir Ferreira 19 Nov '12

19 Nov '12

Hi, I am trying to figure out how much memory a Kamailio server should have to support an X number of users, considering that I am running Kamailio, MySQL and RTPproxy on the same machine. I am planning to use Kamailio at medium/large enterprises, not at service providers but I got the following questions: - Will Kamailio dynamically allocate the system memory it needs or I have to tune up the memory it takes? I.e.: If I use 8G memory, will it take what it needs or I need to adjust it somewhere? - How much memory is reasonable for 10K, 50K and 100K users in such "enterprise" scenarios? - Can you reference any documentation from where I can get this information? Thanks, Moacir

2 1

Kamailio cfg file doubt
by Kamal Palei 17 Nov '12

17 Nov '12

Dear All When I get INVITE (received in UDP), I send it to another proxy in TLS. For this purpose I have modified kamailio.cfg route section , please refer the lines in red. Just before sending in TLS, I modify the IP address in SDP, so that media packets will pass through RTP proxy running When INVITE is reached the next proxy, I see the IP address is changed . So far so good. Next I am trying to modify the IP address in 200 OK to INVITE. However it is not taking effect. I hope we need to modify in onreply_route function. Something is wrong here. I am receiving 200 OK for INVITE in TLS. Next I am sending 200 OK for INVITE to endpoints using UDP. Not sure if onreply_route is being executed or not. Can somebody give me pointer whats wrong here. *# 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])") { force_rtp_proxy("r"); } if (isbflagset(FLB_NATB)) { fix_nated_contact(); } #!endif } * * # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); setflag(FLT_NATS); # NAT detection route(NAT); if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } if(is_method("INVITE|BYE|CANCEL|SUBSCRIBE|REFER|NOTIFY")) { xdbg("incoming request\n"); route(RTPPROXY); t_relay_to("tls:115.114.48.75:443"); exit(); } *Thannks Kamal

2 2

segfault when phone registers on TLS
by Denis 15 Nov '12

15 Nov '12

Hi, I am using kamailio-3.3.2 on Ubuntu 11.04 x64 and when trying to connect via TLS from my Yealink phone kamailio crashes with the following backtrace: Loaded symbols for /lib/x86_64-linux-gnu/libnss_files.so.2 Core was generated by `/opt/kamailio/sbin/kamailio'. Program terminated with signal 11, Segmentation fault. #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 3575 if (unlikely(p->unix_sock<=0)){ (gdb) bt #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 #1 0x000000000051e5bc in send2child (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:3975 #2 handle_tcpconn_ev (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:4310 #3 0x0000000000527dbc in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x00000000004726f5 in main_loop () at main.c:1727 #6 0x000000000047402e in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2546 (gdb) The manual I am following is http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates Thanks, Den

3 10

Kamailio as SMTP server
by Krishna Kurapati 15 Nov '12

15 Nov '12

Hi, Can Kamailio act as SMTP server to receive emails and forward those to SIP users as MESSAGEs? It would be nice if postfix or some other rich SMTP server could function as module of kamailio using all the services provided by kamailio scripting. Krish Kura

2 1

Problem routing to voicemail
by Christophe ROY 15 Nov '12

15 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. My kamailio.cfg is available here as it's too big for the list: http://pastebin.com/zhncJ6j1 What could be the problem? Thank you very much for your help Christophe

1 0

Replicate dialog state
by Kristofer Signer 15 Nov '12

15 Nov '12

In my current setup I've register messages replicated to a second server. Is it possible to also replicate other messages so I can hold dialog state on both servers? I've tried to read the documentation and also google but can't find any examples. Thanks ./Kristofer

2 1

RPC, XMLRPC or other
by Kristofer Signer 14 Nov '12

14 Nov '12

Hello, I'm trying to dig in to the kamailio XMLRPC interfaces and the limitations I read in http://www.kamailio.org/docs/modules/stable/modules/xmlrpc.html#xmlrpc.impl… is really a no go for us. For example, when to retrieve user locations and list of dialogs we get a xml document which is not so well formatted and we can not successfully parse it in an xmlrpc-parser. Is there other options for xmlrpc module? thanks ./Kristofer

4 11

check if the sent reply was local
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, what is the command to check if the reply was recieved or localy generated? Uri

3 3

Re: [SR-Users] sr-users Digest, Vol 90, Issue 39
by Yufei Tao 14 Nov '12

14 Nov '12

What I did to overcome this problem was writing a bash script, that checks if mysql is running, if so, start Kamailio, if not, start mysql first, then start Kamailio, and retries if things fail. And in /etc/rc.local, call this script. Seems to work fine on Ubuntu 10.04. Not tried 12.04 yet. Yufei > > > ------------------------------ > > Message: 2 > Date: Tue, 13 Nov 2012 18:15:05 -0500 > From: Moacir Ferreira <moacirferreira(a)hotmail.com> > Subject: Re: [SR-Users] Returning problem > To: Kamailio Mail List <sr-users(a)lists.sip-router.org> > Message-ID: <COL125-W3986EA29065F0966C8AE6C86C0(a)phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com > To: sr-users(a)lists.sip-router.org > Date: Mon, 12 Nov 2012 14:42:31 -0200 > Subject: Re: [SR-Users] Returning problem > > This is more a system initialization problem, so, take a look in your > /etc/rc<run-level>.d directory (or the correspondent one), specially in > the S* files and find out the start sequence. There You'll find the answer. > > Edson. > > Em 09/11/2012 17:03, Moacir Ferreira escreveu: >> My kamailio start script has the following BEGIN INIT INFO: >> >> # >> ### BEGIN INIT INFO >> # Provides: kamailio >> # Required-Start: $syslog $network $local_fs $time mysql >> # Required-Stop: $syslog $network $local_fs mysql >> # Default-Start: 2 3 4 5 >> # Default-Stop: 0 1 6 >> # Should-Start: postgresql mysql radius >> # Should-Stop: postgresql mysql radius >> # Short-Description: Start the Kamailio SIP proxy server >> # Description: Start the Kamailio SIP proxy server >> ### END INIT INFO >> >> However, when I boot my server Kamailio will start before mysql. So, >> after boot I can start Kamalio but at boot time it will fail. >> >> Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. >> All the code has been compiled at the server. >> >> Cheers! >> >> Moacir >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > > >

1 0

update the to_tag on a local generated reply
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, When i set a timer to "request time out" kamilio generates a 408 with now to_tag. I saved the to_tag from the 183 and try to update the acc table with the to_tag (i need that for my CDR to work) My question is (understanding there is no option to set the to_tag on the 408) is: How to i get the right timing to update using sqlops? i must update after the acc inserted the row to the acc table.... Any ideas? Thanks, Uri

2 3

Presence Dialog Info synchronisation issue
by David 12 Nov '12

12 Nov '12

Hello, I have two Kamailio boxes running. First one routes calls between my phones and asterisk. Here are the extra lines compared to the default config that came with my installation on the first server. Let's call this server kamailio : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 0) modparam("pua_dialoginfo", "include_tags", 0) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua", "outbound_proxy", "sip:kamailio_presence_ip:5060") if ( is_method("SUBSCRIBE") ) { t_relay("kamailio-presence-ip", "5060"); } The second one handles presence stuff ( SUBSCRIBE and PUBLISH ), let's call it presence : After the standard sanity checks : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; The problem is that I have two phones connected to the same username on the server. Here is the sequence of events of a correctly functionning call : 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Presence replies with an OK, which includes a SIP-ETag header 6. Kamailio recieves the OK 7. Kamailio receives the 180 Ringing from the second phone 8. Kamailio sends a new PUBLISH with the SIP-If-Match using the value from the SIP-ETag header 9. Presence receives it, updates the presence table and sends an OK. This is ok, the trouble is when the second 180 Ringing reaches Kamailio before the OK to the publish with the SIP-ETag. 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Kamailio receives the 180 Ringing from the second phone 6. Kamailio sends a PUBLISH to presence 7. Presence replies with an OK, which includes a SIP-ETag header 8. Presence replies with an OK, which includes a second SIP-ETag header 9. Kamailio recieves the OK 10. Kamailio recieves the OK The presence table now has two seperate dialogs for the call because Kamailio didn't send the SIP-If-Match header as it did not yet receive the reply from presence. When the two phones start ringing at the same time, the time between the two packets arriving at the server is shorter than a round trip to the presence server. In the second scenario, Kamailio never updates the presence information using the first SIP-ETag, so the light on my phone stays stuck at "ringing". Where do I go from here ? Thanks, David

1 0

Returning problem
by Moacir Ferreira 12 Nov '12

12 Nov '12

My kamailio start script has the following BEGIN INIT INFO: # ### BEGIN INIT INFO # Provides: kamailio # Required-Start: $syslog $network $local_fs $time mysql # Required-Stop: $syslog $network $local_fs mysql # Default-Start: 2 3 4 5 # Default-Stop: 0 1 6 # Should-Start: postgresql mysql radius # Should-Stop: postgresql mysql radius # Short-Description: Start the Kamailio SIP proxy server # Description: Start the Kamailio SIP proxy server ### END INIT INFO However, when I boot my server Kamailio will start before mysql. So, after boot I can start Kamalio but at boot time it will fail. Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. All the code has been compiled at the server. Cheers! Moacir

2 1

LUA 5.2 - Support
by David | StyleFlare 12 Nov '12

12 Nov '12

Curious to know if there is any plans to support LUA 5.2. Seems like this is a newer release, I tried to build kamailio lua modules using 5.2 but it was not supported. Seems like some minor changes in the way LUA handles some things. Any ideas? Thanks.

2 1

xcap: no more space for spec value
by Juha Heinanen 12 Nov '12

12 Nov '12

i got the these error messages when my xcap server tried to serve 'GET xcap-caps' request: Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: INFO: XCAP service GET <xcap-caps> Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1305]: no more space for spec value Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1314]: buffer overflow -- increase the buffer size... Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [sr_module.c:1665]: Could not convert the PV-formated string to str Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: xhttp [xhttp_mod.c:461]: unable to get body config code where this comes is this: xlog("L_INFO", "XCAP service $rm <$xcapuri(u=>auid)>\n"); $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); i have not checked yet, which buffer should be increased, but would it be a good idea to make the default bigger? -- juha

2 2

Use different flags in rtpproxy_manage() based on from which route its called
by Aft nix 12 Nov '12

12 Nov '12

Hi, rtpproxy_manage() combines both rtpproxy_offer() and rtpproxy_answer(). Now if i want different flags set for rtpproxy_offer() and rtpproxy_answer() then how should i use rtpproxy_manage()? To specific, what i want : route[REQUEST]{ ..... rtpproxy_offer("z20"); ... } onreply_route[]{ .... rtpproxy_answer("z120"); .... } Now in kamailio.cfg there is one call rtpproxy_manage() in route[NATMANAGE]. route[NATMANAGE] is called from both route[REQUEST] and onreply_route[]. So i can't emulate the functionality described above. My question is using only rtpproxy_manage() is it possible to emulate the above functionality? Thanks in advance. -- -aft

3 3

Homer multiple domains
by Reda Aouad 11 Nov '12

11 Nov '12

Hi I am using homer and managing multiple SIP domains. My problem is that in the sip_capture table, the FROM and TO domains are not logged in the FROM and TO fields. The MSG field contains the domains, but the queries involving domains become tricky and heavy to execute. Is there a way I can have the domains in their respective FROM/TO columns? Thanks Reda

1 0

how to use "t_drop_replies" but keep the last reply for acc
by Uri Shacked 11 Nov '12

11 Nov '12

Hi, I am trying to make an option of "route when no answer" or " route when busy". What I am doing is checking the reply and if "busy", for example, I use "t_drop_replies". Then, I set the new number and route[relay] again. On the accdb table, I get the first invite with 183 and after that the second invite with 183 and with 200. I would like to do exactly what i do, but would like to see on the accdb the 486 reply from the first invite. how do i do it? BR, Uri

4 11

Kamailio language definitions- dictionary
by kellon＠sis-grenada.com 11 Nov '12

11 Nov '12

Hi All Can anyone direct me to some go literature/documentation on the language/functions used in the kamailio config file. E.g cfg_get(), strempty(), proxy_authorize(), sl_send_reply() etc.... Thanks Kellon

2 1

No Debug Log message
by kellon＠sis-grenada.com 10 Nov '12

10 Nov '12

Hi all I am currently having to modify the kamailio config file without the benefit of log messages in /var/log/message. I am using the default configuration parameter settings of: #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 Has anyone experience this issue before or have any suggestions on howto fix. Making changes to the config file line by line and restarting the daemon is killing me!!!

2 1

Kamailio CDR logging to RADIUS
by Owen Lynch 10 Nov '12

10 Nov '12

Hi, I have transaction accounting to RADIUS setup and get the start and stop events logged to file when a call is made. I want to extend this to also get CDR logs - call duration etc. To do this I understand I need to configure the acc module and load and configure the dialog module: #!define FLD_DLG 4 modparam("acc", "cdr_enable", 1) modparam("dialog", "dlg_flag", FLD_DLG) I was expecting extra entries in the RADIUS file but they don't appear. I think this is because I need to set and clear the dialog flag at appropriate places in the route script (e.g. setflag(FLD_DLG)) but I'm not sure where. Should this be done in the INVITE and BYE method handling as is currently done for the transaction accounting, e.g.: # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } Thanks, Owen Lynch

3 4

"branch" tag in the "Via" header of the ACK message for the re-INVITE
by Alex Solt 10 Nov '12

10 Nov '12

Hi, We have a problem with the "branch" tag in the "Via" header of the ACK message for the re-INVITE. The "branch" tag value is not a unique number so the end point thinks the ACK is belong to the previous call. This causes the active calls fail after 15 min (right after the ACK message for re-INVITE passed by kamailio proxy). Is there a way to set the "branch" tag in the "Via" header of the ACK message for the re-INVITE to be a unique number? PS1: We are running kamailio v3.3.PS2: we did not have this issue with the old version of openser. Thanks,AS

2 3

Dispatcher question
by Mino Haluz 10 Nov '12

10 Nov '12

Hi, I have simple dispatcher that is dispatching initial invites to the core router. The question is, how can I tell kamailio to not add the Via header in the INVITE. I mean, I want that this dispatcher will be completely transparent to the core router, so any responses will go directly to the original sender, not to the dispatcher. This is my request route: request_route { if (is_method("INVITE")) { if (!ds_select_domain(1, 0)) { xlog("L_INFO", "[MAIN] ERROR: Proxy1 failed"); if (!ds_select_domain(2, 0)) { xlog("L_INFO", "[MAIN] FATAL: Proxy2 failed"); exit; } } xlog("L_INFO","[MAIN] Dispatching INVITE to $rd"); forward(); } } 1 and 2 are defined in dispatcher.list. If it can be done in kamailio, please how. Thank you! Mino

3 5

presence module makes excessive cleanups when subs_db_mode=3
by Andrew Pogrebennyk 09 Nov '12

09 Nov '12

Hi, when presence module is running with subs_db_mode=3 it makes an excessive number of SQL select queries, litelly dozens per second: > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event='presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event<>'presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event='presence.winfo' Steps to reproduce: registers two subscribers in jitsi and add them to contact lists of each other. The kamailio version is 3.3.2. I'm not doing anything special configuration-wise: loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") # in 3.3 the fallback2db change to subs_db_mode modparam("presence", "subs_db_mode", 3) modparam("presence", "notifier_processes", 3) loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) # retry-after 5 minutes modparam("presence_xml", "xcapauth_userdel_reason", "probation;retry-after=300") Q: it is a bug of a feature? :) Thanks. Andrew

2 2

xmlrpc dlg.end_dlg problem
by Kristofer Signer 09 Nov '12

09 Nov '12

Hi all, I have an kamailio 3.3.2 server deployed with dialog awareness. When call dlg.end_dlg over xmlrpc I get an 200 OK response but the dialog is not ended. What I can do; call dlg.list (xmlrpc) and get the active dialog list, xmlrpc interface working. call dlg.dlg_list (xmlrpc) and get description of one dialog, xmlrpx method call with parameters working. call dlg_end_dlg (kamctl) and successfully end the dialog. so my questions are... Is the xmlrpc method dlg.end_dlg known to work? and if, what have I missed? thanks ./Kristofer Signer $ /usr/local/kamailio-3.3/sbin/kamailio -V version: kamailio 3.3.2 (x86_64/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, select, kqueue. id: unknown compiled on 09:53:26 Nov 2 2012 with gcc 4.2.1

2 3

dispatcher: What's "first destination" in a database
by Olle E. Johansson 08 Nov '12

08 Nov '12

Friends, Dispatcher has a mode 8 - "use first destination" which makes sense for a textfile driven dispatcher configuration. Now if I use a database - will dispatcher first sort on priority, then take the first? It seems so from the documentation of the text file: priority: sets the priority in destination list (based on it is done the initial ordering inside the set) Which one is top priority - lowest or highest value? I can update the docs to clarify if someone helps me with the answer :-) /O

2 1

Another NAT question
by Moacir Ferreira 08 Nov '12

08 Nov '12

Hi, I have a scenario where I got a firewall with 3 interfaces: internal, DMZ and external. All the traffic from internal going to external is NATed. However, the traffic between internal and DMZ is NOT NATed. The external and DMZ are using public IP addresses. On the DMZ I have a Kamailio server running with RTPProxy + NAT Helper. Everything works fine when public (from the internet) users (UAs) are behind NAT as Kamailio will force the RTP to go via RTPProxy. However, when the public user has a public IP, it will fail to establish the RTP to a user who registered on Kamailio coming from the internal firewall interface. The reason is that, as I don't do NAT between internal and DMZ firewall interfaces, Kamailio will not RTPRroxy in between the UAs because, from the way Kamailio detects NAT, they are not behind NAT. So the public user UA tries to reach a private IP address. If the "inside" user tries to call a public user (a user on the Internet with a public IP), it works. Yes, I could do NAT in between the internal and DMZ firewall interfaces. However, this would force all RTP traffic of my UAs at the LAN go to Kamailio RTPProxy, an bad effect that I would like to avoid. So my question is: what would be the best approach to solve this issue using Kamailio and RTPProxy in such scenario? Cheers! Moacir

2 4

siptrace buffer
by Martin Formanko 08 Nov '12

08 Nov '12

Hi, do you know what happens if the database for siptrace is not reachable ? I assume siptrace is non-blocking in new versions of kamailio, but is there any buffer? Or insert queries are discarded when timed out ? Mino

2 1

Gateway failover on slow call setup
by Spencer Thomason 08 Nov '12

08 Nov '12

Hello, I'm trying to address a problem where sometimes our upstream providers have slow call setup times. We use drouting with multiple providers for each route to handle failover in the event of an outage but in this case, we get a 100 Trying but sometimes a 18[03] response takes 4-5 seconds or we never get it because the caller cancels the request. What would be a "safe" amount of time to allow a PSTN gateway to reply with a 180/183? What I'm trying to avoid is a scenario where the PSTN legitimately takes a long time to setup (i.e. an international call to some cell phone providers) and we needlessly failover which will actually increase the post dial delay for the user. My thought is this for example: route[PSTN] { t_set_fr(1000); # set fr timer to 1 second so we failover quickly if no provisional response t_on_reply("PSTN_REPLY"); # Routing logic } onreply_route[PSTN_REPLY] { if (status=~"1[0-9][0-9]") { t_set_fr(4000); # set fr timer to 4 seconds } if (status=~"18[03]") { t_reset_fr(); # reset timer if we get a 180 or 183 } } Also, if we failover because of slow call setup for a particular route, I'd like to temporarily avoid that gateway for that route (but only that route). I'm curious if anyone has any pointers or how they solved this. Thanks, Spencer

2 1

Adding Users not excepting email option
by kellon＠sis-grenada.com 08 Nov '12

08 Nov '12

I have attempted to add users as per documentation. kamctl add <username> <password> <email> However, Kamailio responds with the following: -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) Is the option to add user with email address etc still available in the current release of 3.3 Thanks Kellon

2 1

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

Thanks Anton, Fred, Jeremy thanks for the suggestions. The problem was the firewall. I don't know why I didn't think of it. I learnt one other thing also the use of kamctl monitor. Just two other problems at the moment. I'll start a new thread for them

1 0

TLS Certificate Verification Issue
by Kamal Palei 07 Nov '12

07 Nov '12

Dear All I have modified kamailio,cfg and compiled all the modules with TLS enabled, and able to bring up the kamailio proxy properly. Kamailio proxy will receive the REGISTER message from endpoints in UDP , and want to send this REGISTER message to another intermediate proxy in TLS. For this purpose, I have added few lines in kamailio.cfg file as below. I have created the certificates, private keys as explained by README file in kamailio-3.1.5/modules/tls/ path. if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } Looks like this is taking effect. When Kamailio receives REGISTER message it is trying to do handshake with intermediate proxy. I used wireshark to see the handshake messages. 1. From Kamailio proxy, a TCP SYNC message is going to intermediate proxy. 2. intermediate proxy sends SYNC + ACK 3. Kamailio sends CLIENT HELLO 4. intermediate proxy sends SERVER HELLO, CERTIFICATE and SERVER HELLO DONE 5. The Kamailio sends ALERT (Level: Fatal, Description: Unknown CA) ---> IS something going wrong here.............. 6. Then Kamailio sends FIN + ACK Can somebody please let me know why the certificate verification fails (I get this log in console). How can I put a work around to avoid certification verification failure. Best Regards kamal

2 10

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users