- sr-users - kamailio.org

Debs for Kamailio v4.1.0
by Daniel-Constantin Mierla 06 Dec '13

06 Dec '13

Hello, hopefully debs will became available soon to deb.kamailio.org, as Jon Bonilla will get a chance to prepare the build system for this new v4.1.0. Meanwhile, there are some alternative builds for Debian 6.0, 7.0 and Ubuntu 12.04 at: - https://build.opensuse.org/package/show/home:kamailio:telephony-debs/kamail… - https://build.opensuse.org/package/show/home:kamailio:telephony-debs/kamail… - https://build.opensuse.org/package/show/home:kamailio:telephony-debs/kamail… Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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RPM packages for Kamailio v4.1.0
by Daniel-Constantin Mierla 06 Dec '13

06 Dec '13

Hello, rpm packages for quite a bunch of Linux-based distributions (e.g., Centos, Fedora, OpenSuse) are available at: - https://build.opensuse.org/package/show/home:kamailio:telephony/kamailio41 Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

missing mediaprpxy-ng debian dependency
by Juha Heinanen 05 Dec '13

05 Dec '13

i tried to install mediaproxy-ng by apt-get and got the following error: Error: Your kernel headers for kernel 3.2.0-4-486 cannot be found. Please install the linux-headers-3.2.0-4-486 package. apt-get automatically installed linux-headers-3.2.0-4-common and linux-headers-3.2.0-4-686-pae, but not the above mentioned package. is there some dependency (like linux-headers-3.2.0-4-all) missing from debian/control file? -- juha

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Kamailio- Redirect message 300 with LCR reroute
by Fatima Chahrour~Vanrise Support 05 Dec '13

05 Dec '13

Dear List, I'd like to get help regarding my case. I have the following script where many thanks to Daniel has helped me in if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} while(next_gw()){ km_append_branch(); } sl_send_reply("302","Moved Temporary"); exit; } The problem am facing is that the call is rerouting from the first gateway to the next gateway successfully when I have two gateways. BUT when I have three gateways, and the first two gateways are off, the call is not rerouted to the third gateway. It keeps hitting the seconds gateway and gives request time out at the end Why is that? Thanks in advance, F Chahrour

3 14

kamailioi is not relaying BYE message to UAC
by Arif Hossain 05 Dec '13

05 Dec '13

Hi, We have the following network architecture : UAC1------------------------->kamailio-------------------->VoipSwitch----->PSTN---------->Phone1 (Sip Client) Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session is terminated cleanly. But if Phone1 hangs up the BYE message which comes to kamailio and goes back to VoipSwitch instead of relayed to UAC1 . So The session becomes a zombie one, And UAC1 unfortunately gets billed for a session which should be terminated. Following is the Call flow as seen from VoipSwitch : | <kamilio IP> | | | | <VoipSwitch IP> | |134.856 | INVITE SDP | |(7890) ------------------> (5060) | |134.858 | 407 Proxy Authentication Required | |(7890) <------------------ (5060) | |134.902 | ACK | | |(7890) ------------------> (5060) | |135.408 | INVITE SDP | |(7890) ------------------> (5060) | |135.414 | 100 Trying| | |(7890) <------------------ (5060) | |140.121 | 183 Session Progress SDP | |(7890) <------------------ (5060) | |140.184 | RTP (g729) | | |(61868) <------------------ (5136) | |141.295 | RTP (g729) | | |(61868) ------------------> (5136) | |153.701 | 200 OK SDP | |(7890) <------------------ (5060) | |153.713 | RTP (g729) | | |(61868) ------------------> (5136) | |154.126 | ACK | | |(7890) ------------------> (5060) | |159.988 | BYE | | |(7890) <------------------ (5060) | |160.031 | BYE | | |(7890) ------------------> (5060) | |160.478 | BYE | | |(7890) ------------------> (5060) | |161.412 | BYE | | |(7890) ------------------> (5060) | |181.952 | BYE | | |(7890) ------------------> (5060) | |185.687 | BYE | | |(7890) ------------------> (5060) | |188.018 | 408 Request Timeout | |(7890) ------------------> (5060) | Sip Traces : kamailio------>VoipSwitch I'm posting only the offending BYE msg instead of full trace , because of the mail will become difficult to read . If more traces needed, i can post it. The following BYE message is sent by VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Route: <sip:108.166.195.189:7890;lr=on;nat=yes> CSeq: 2 BYE Via: SIP/2.0/UDP 205.164.40.74:5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" <sip:ipphone@205.164.40.74>;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 Kamailio instead of relaying the message, sends a BYE message towards VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Max-Forwards: 10 CSeq: 2 BYE Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" <sip:ipphone@205.164.40.74>;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 When the first BYE message comes from VoipSwitch , kamailio does the following : May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:753]: sdp = 0x83043dc May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:755]: session = 0x8304504 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:632]: method: <BYE> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:ipphone@205.164.40.74> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <BYE> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bk19052612230719933454843>; state=16 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:145]: After parse_msg... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=arILprdVR1srJ76HHlt4BEc3XsyaWcZm May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [76]; uri=[sip:ipphone@205.164.40.74] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body ["ipphone" <sip:ipphone@205.164.40.74>] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=1905251223419334290936029 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:85]: is_preloaded: No May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && [205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && [205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [forward.c:448]: check_self: host != me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 15==15 && [108.166.195.189] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 7890 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:792]: Topmost route URI: 'sip:108.166.195.189:7890;lr=on;nat=yes' is me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:257]: No next Route HF found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:811]: No next URI found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:983]: params are <;lr=on;nat=yes> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2501 , global msg id=2500 , T on entrance=0xffffffff May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=29177, isACK=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:564]: DEBUG: proceeding to pre-RFC3261 transaction matching May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb61626a4, callback type 1, id 0 entered May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) May 20 02:26:04 VOS20-108 last message repeated 5 times May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up May 20 02:26:22 VOS20-108 /usr/local/sbin/kamailio[16445]: DEBUG: <core> [udp_server.c:486]: udp_rcv_loop: probing packet received from 180.234.62.230 38722 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1134]: ->>>>>>>>> T_code=0, new_code=408 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1636]: DEBUG: relay_reply: branch=0, save=0, relay=0 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: <core> [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: <core> [mem/shm_mem.c:105]: WARNING:vqm_resize: resize(0) called May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61626a4, callback type 128, id 0 entered May 20 02:26:54 VOS20-108 /usr/local/sbin/kamailio[16451]: WARNING: <core> [timer.c:450]: WARNING: our timer runs faster then real-time (300000 ms / 4800 ticks our time .-> 299923 ms / 4798 ticks real time) OT: How do you guys maintain 80 column mails? i do not use a mail client, use gmail mailbox. -- -aft

5 6

Kamailio v4.1.0 Released - new major version is out
by Daniel-Constantin Mierla 05 Dec '13

05 Dec '13

Kamailio v4.1.0 is out – a new major release, bringing out as usually a very large set of new features and improvements. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-1-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.1.0 and have a great time during winter holidays! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 2

mediaproxy-ng throughput
by Alex Balashov 05 Dec '13

05 Dec '13

So, I hate to be the bearer of the most banal, cliché, overplayed question ever, especially since I am perfectly aware of the formidable challenges in getting the data and the variance in the results, but: Are there any data points on how much of a throughput premium mediaproxy-ng/rtpproxy-ng offers over userspace rtpproxy Classic(TM)? -- 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/

1 0

Ugrade Kamailio 4.0.4 to 4.1.0
by mohammed alyaseen 05 Dec '13

05 Dec '13

Hello I am using kamailio 4.0.4. I want to change to kamailio 4.1.0. Can the version 4.0.4 be upgraded to the new version 4.0.4 or Do I have to uninstall it and install the 4.1.0? Thanks in advance Medo

3 4

407 Proxy Authentication Required
by mohammed alyaseen 05 Dec '13

05 Dec '13

Hi all...I need help ... I am using Kamailio as SIP server... I tried to do call between two sip clients (MyMonster)..The clients can register with kamailio without any problems. But when I try to do call between two clients, It doesn't work. Once the kamailio gets the first invite from the client, it directly replies with "407 Proxy Authentication Required". In the kamailio.cfg the domain is "kamailio.test" in the kamctlr I also use "kamailio.test" as SIP domain.. In the Log of Kamailio I see this: 2(16979) DEBUG: corex [corex_lib.c:189]: corex_check_self(): check self domain match: 0:kamailio.test:0 2(16979) DEBUG: auth_db [authorize.c:476]: auth_check(): realm [kamailio.test] table [subscriber] flags [1] 2(16979) DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm 'kamailio.test' not found 2(16979) DEBUG: auth_db [authorize.c:252]: digest_authenticate(): no credentials 2(16979) DEBUG: auth [challenge.c:127]: get_challenge_hf(): build_challenge_hf: realm='kamailio.test' 2(16979) DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth: 'Proxy-Authenticate: Digest realm="kamailio.test", nonce="Up+EuVKfg42WYOrC87GTm+rvMNFeNe3/BCf5m4A=" Thanks in advance... Medo

2 1

Accounting to mysql resulted in segfault
by Daniel Tryba 05 Dec '13

05 Dec '13

This morning I had the unpleasant realisation that kamailio (4.0.3) crashed twice while I was running an alter statement on the acc table of a live instance. db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on q uery: Too many connections <core> [db_query.c:132]: db_do_query_internal(): error while submitti ng query auth_db [authorize.c:175]: get_ha1(): failed to query database or alias_db [alookup.c:111]: alias_db_lookup(): failed to query database db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Too many connections just before (no core): segfault at 18 ip 00007f0d6b287a83 sp 00007fffc4467550 error 4 in usrloc.so[7f0d6b267000+27000] So instead of just discarding the acc inserts, these pile up resulting in consuming max connections to the database and eventually segfaulting. I know (now) that I shouldn't do this on a production server, but I'd rather have the database accounting discarded than a machine crashing (can be reconstructed from syslog). My question: is this "normal" behavior? Can this be avoided somehow (other than ensuring 100% availability of the sql server). -- 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 2

reload to memory is much slower and has problms after upgrade to 3.3.2 and using mem_join=1
by Uri Shacked 05 Dec '13

05 Dec '13

Hi, I am using MTREE and DIALPLAN modules to load lots of info to kamailio. (6 million rows). When kamailio was running with 3.2.1 (no mem_join=1 option), the used size was increasing but the process of loading the data was fast eanough. I upgraded to 3.3.2 and set mem_join=1. Now the loading process take about 10 time longer and sometimes stops kamailio from responding to traffic. Any ideas? Thanks, Uri

3 19

Re: [SR-Users] Logging and debugging
by Keith 05 Dec '13

05 Dec '13

Thanks for the quick reply, i'll have a look at that! I am also keen to log SIP invite errors etc, is this normally shown in the CDR table? Cheers Keith

1 0

4.0.4 SEGFAULT again
by Kelvin Chua 05 Dec '13

05 Dec '13

the previous segfault was fixed, now this is a different, i cannot replicate this consistently. randomly crashes. full backtrace: #0 qm_detach_free (frag=0x27d926b4da0, qm=<optimized out>) at mem/q_malloc.c:269 prev = 0x27d923ad0a8 next = 0x0 #1 qm_malloc (qm=0x27d923ad000, size=48) at mem/q_malloc.c:386 f = 0x27d926b4da0 hash = -1841639256 #2 0x0000027d99431c76 in insert_tmcb () from /usr/local/lib64/kamailio/modules/tm.so No symbol table info available. #3 0x0000027d96c0420b in dlg_set_tm_callbacks () from /usr/local/lib64/kamailio/modules/dialog.so No symbol table info available. #4 0x0000027d96c04508 in dlg_onreq () from /usr/local/lib64/kamailio/modules/dialog.so No symbol table info available. #5 0x0000027d9943101b in ?? () from /usr/local/lib64/kamailio/modules/tm.so No symbol table info available. #6 0x0000027d9940f052 in build_cell () from /usr/local/lib64/kamailio/modules/tm.so No symbol table info available. #7 0x0000027d9943bfc6 in t_newtran () from /usr/local/lib64/kamailio/modules/tm.so No symbol table info available. #8 0x0000027d99420cc3 in t_relay_to () from /usr/local/lib64/kamailio/modules/tm.so No symbol table info available. #9 0x000000000041d58b in do_action (h=0x3ff745086c0, a=0x27d9a40c170, msg=0x27d9a458be0) at action.c:1074 ret = -5 v = <optimized out> dst = {send_sock = 0x3ff74507b70, to = {s = {sa_family = 35808, sa_data = "E\232}\002\000\000\370{Pt\377\003\000"}, sin = {sin_family = 35808, sin_port = 39493, sin_addr = {s_addr = 637}, sin_zero = "\370{Pt\377\003\000"}, sin6 = { sin6_family = 35808, sin6_port = 39493, sin6_flowinfo = 637, sin6_addr = {__in6_u = { __u6_addr8 = "\370{Pt\377\003\000\000\340\213E\232}\002\000", __u6_addr16 = {31736, 29776, 1023, 0, 35808, 39493, 637, 0}, __u6_addr32 = {1951431672, 1023, 2588249056, 637}}}, sin6_scope_id = 6372243}}, id = 0, proto = -77 '\263', send_flags = { f = 138 '\212', blst_imask = 74 'J'}} tmp = <optimized out> new_uri = <optimized out> end = <optimized out> crt = <optimized out> cmd = 0x27d9a3eddd8 len = <optimized out> user = <optimized out> uri = {user = {s = 0x3ff74507810 "H", len = -1706983472}, passwd = { s = 0x27d97473a60 "\220\202E\232}\002", len = -1759063584}, host = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = 44882432}, port = { s = 0x27d9a3e6be0 "gw::current", len = 11}, params = {s = 0x1 <Address 0x1 out of bounds>, len = -1706983328}, sip_params = {s = 0x3ff74507810 "H", len = -1706718240}, headers = { s = 0x3ff74507b40 "\001", len = -1}, port_no = 39664, proto = 39492, type = 637, flags = 4885171, transport = {s = 0x3ff74508050 "\037\067c", len = 44882432}, ttl = { s = 0x3ff74507b40 "\001", len = 1951431488}, user_param = {s = 0x3ff745086c0 "\003", len = 5148184}, maddr = {s = 0x63371f "", len = -1759094110}, method = { s = 0x27d9a449af0 "0\265?\232}\002", len = 4310469}, lr = {s = 0x27d9a458be0 "F]8", len = -1706761360}, r2 = {s = 0x48 <Address 0x48 out of bounds>, len = -1841596304}, gr = { s = 0x3ff74507890 "@x>\232}\002", len = -1706720624}, transport_val = {s = 0x27d926121a0 "", len = -1707182016}, ttl_val = {s = 0x27d9a449af0 "0\265?\232}\002", len = -1706968808}, user_param_val = {s = 0x27d9a41b918 "\002", len = -1706967088}, maddr_val = { s = 0x27d9a458be0 "F]8", len = 1951434432}, method_val = {s = 0x3ff74507b50 "\037\067c", len = 5170540}, lr_val = {s = 0x0, len = -1713638448}, r2_val = { s = 0x27d9a3e7840 "gw::current", len = 11}, gr_val = {s = 0x0, len = -1706968832}} next_hop = {user = {s = 0x3ff74507b68 "\305\305A", len = -1706718240}, passwd = { s = 0x3ff745086c0 "\003", len = 1951431528}, host = {s = 0x3ff74507b20 "", len = 4885171}, port = {s = 0x13 <Address 0x13 out of bounds>, len = -1706968848}, params = { s = 0x3ff74507b40 "\001", len = -1706718240}, sip_params = {s = 0x3ff745086c0 "\003", len = 5165842}, headers = {s = 0x27d92804290 "\020", len = 5746748}, port_no = 32448, proto = 29776, type = 1023, flags = 0, transport = {s = 0x27d92b1a090 "\002", len = -1765734391}, ttl = {s = 0x0, len = 0}, user_param = { s = 0x3ff74507b60 "\360\232D\232}\002", len = 1951431520}, maddr = { s = 0x800000000 <Address 0x800000000 out of bounds>, len = 9689456}, method = { s = 0x3ff74507a70 "", len = 9598498}, lr = {s = 0x2 <Address 0x2 out of bounds>, len = -1833852784}, r2 = {s = 0x0, len = -1706718240}, gr = {s = 0x27d9a41b8f8 "\a", len = -1765730574}, transport_val = {s = 0x27d96e3c890 "", len = -1765795157}, ttl_val = { s = 0xffffffff00000000 <Address 0xffffffff00000000 out of bounds>, len = -1765722972}, user_param_val = {s = 0x27d98330d4a "<script>: ", len = -1713638448}, maddr_val = { s = 0x27d9a456f00 "@", len = 5746748}, method_val = {s = 0x27d9a469df8 "\006", len = -1707093008}, lr_val = {s = 0x27d9a458be0 "F]8", len = 44882432}, r2_val = { s = 0x27d9a458be0 "F]8", len = -1706965328}, gr_val = {s = 0x27d9a458be0 "F]8", len = 1951434432}} u = <optimized out> port = <optimized out> dst_host = <optimized out> i = <optimized out> flags = <optimized out> avp = <optimized out> st = {flags = 1951432672, id = 1023, name = {n = 0, s = {s = 0x0, len = -1706779920}, re = 0x0}, avp = 0x4ee90f} sct = <optimized out> sjt = <optimized out> rve = <optimized out> mct = <optimized out> rv = <optimized out> rv1 = <optimized out> c1 = {cache_type = 2588249056, val_type = 637, c = {avp_val = {n = 1, s = { s = 0xc00000001 <Address 0xc00000001 out of bounds>, len = 0}, re = 0xc00000001}, pval = { rs = {s = 0xc00000001 <Address 0xc00000001 out of bounds>, len = 0}, ri = -1765782691, flags = 637}}, i2s = "\001\000\000\000\000\000\000\000\244\064\301\226}\002\000\000\037\067c\000\000"} s = {s = 0x1 <Address 0x1 out of bounds>, len = -1765690355} srevp = {0x92763d, 0x27d96c134a4} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NUMBER_ST, u = {number = 2, string = 0x2 <Address 0x2 out of bounds>, str = {s = 0x2 <Address 0x2 out of bounds>, len = 0}, data = 0x2, attr = 0x2, select = 0x2}}, {type = RVE_FREE_FIXUP_ST, u = { number = 0, string = 0x0, str = {s = 0x0, len = 5}, data = 0x0, attr = 0x0, select = 0x0}}, { type = RVE_FREE_FIXUP_ST, u = {number = 0, string = 0x0, str = {s = 0x0, len = 4}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = { s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = { number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = { s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #10 0x000000000041c5c5 in run_actions (h=0x3ff745086c0, a=0x27d9a405e80, msg=0x27d9a458be0) at action.c:1573 t = <optimized out> ret = <optimized out> mod = <optimized out> ms = 0 __FUNCTION__ = "run_actions" #11 0x000000000041daaa in do_action (h=0x3ff745086c0, a=0x27d9a4498f0, msg=0x27d9a458be0) at action.c:690 ret = -5 v = <optimized out> dst = {send_sock = 0x27d9969e080, to = {s = {sa_family = 0, sa_data = "\000\000\000\000\000\000\037\067c\000\000\000\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\037\067c\000\000\000\000"}, sin6 = { sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "\037\067c\000\000\000\000\000\001\000\000\000\f\000\000", __u6_addr16 = { 14111, 99, 0, 0, 1, 0, 12, 0}, __u6_addr32 = {6502175, 0, 1, 12}}}, sin6_scope_id = 2453371216}}, id = 637, proto = -94 '\242', send_flags = {f = 163 '\243', blst_imask = 67 'C'}} tmp = <optimized out> new_uri = <optimized out> end = <optimized out> crt = <optimized out> cmd = <optimized out> len = <optimized out> user = <optimized out> uri = {user = {s = 0x27d9ab95fb0 "\200\235\024\002", len = 1951432016}, passwd = { s = 0x27d9ab95720 "", len = 34928992}, host = {s = 0x109 <Address 0x109 out of bounds>, len = 44882432}, port = {s = 0x120 <Address 0x120 out of bounds>, len = -1706721952}, params = {s = 0x1 <Address 0x1 out of bounds>, len = 0}, sip_params = {s = 0x63371f "", len = 100}, headers = {s = 0x937560 "", len = 4310469}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = URI_USER_NORMALIZE, transport = { s = 0xfffffffb <Address 0xfffffffb out of bounds>, len = 44882432}, ttl = { s = 0x15 <Address 0x15 out of bounds>, len = -1707088000}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x63371f "", len = -1}, method = {s = 0x27d923b7950 "`\315b\222}\002", len = 4310469}, lr = {s = 0x27d9a458be0 "F]8", len = -1706721952}, r2 = { s = 0x7100000000 <Address 0x7100000000 out of bounds>, len = -1706721696}, gr = { s = 0x1 <Address 0x1 out of bounds>, len = 0}, transport_val = {s = 0x63371f "", len = 100}, ttl_val = {s = 0x937560 "", len = 4315438}, user_param_val = {s = 0x3ff74507e70 "\001", len = 1951432192}, maddr_val = {s = 0x27d9a458be0 "F]8", len = -1707088000}, method_val = { s = 0x1 <Address 0x1 out of bounds>, len = 1951434432}, lr_val = {s = 0x0, len = 1951434432}, r2_val = {s = 0x3ff74508530 "\001", len = -1}, gr_val = {s = 0x27d923b7950 "`\315b\222}\002", len = 4346308}} next_hop = {user = {s = 0x0, len = 1}, passwd = {s = 0x3ff745086c0 "\003", len = -1671675758}, host = {s = 0x3ff745086c0 "\003", len = 1951434032}, port = { s = 0xffffffff <Address 0xffffffff out of bounds>, len = -1841596080}, params = { s = 0xe8d81dd966fc4092 <Address 0xe8d81dd966fc4092 out of bounds>, len = 154943634}, sip_params = {s = 0x3ff00000000 <Address 0x3ff00000000 out of bounds>, len = 8187}, headers = { s = 0x1 <Address 0x1 out of bounds>, len = 1}, port_no = 1, proto = 0, type = ERROR_URI_T, flags = 11, transport = {s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, .... cut off Kelvin Chua

1 0

Logging and debugging
by Keith 04 Dec '13

04 Dec '13

Hi guys, Does anyone have any advice to good Kamailio logging and debugging? When you switch on debugging the file grows quickly and with information that isn't really required. My initial idea is to run through the code and make lots of comments but just wanted other peoples ideas/views. Thanks, as always. Keith

2 1

Re: [SR-Users] COR
by Keith 04 Dec '13

04 Dec '13

Thanks, but I need to restrict based on registered user, i.e. some users will be able to dial those numbers others wont. Thanks, Keith

2 1

Considering to release v4.1.0 on Dec4, 2014
by Daniel-Constantin Mierla 04 Dec '13

04 Dec '13

Hello, I have few issues on my list to fix for 4.1 before the full release, but I consider one week being enough time. Therefore I plan to release 4.1.0 next Wednesday, Dec 4, 2014. If anyone would prefer a different plan, expose it on the mailing list for further discussions. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 4

IPv4, IPv6, RTPProxy and Kamailio
by Mark Zeman 04 Dec '13

04 Dec '13

Hello all, The subject says most of it, I think. We set up our Kamailio and RTPProxy according to http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 with the addition of an alias (siplab.ch), and the DNS to go with it, as well as TLS and SRTP. However, we only get working calls IPv4-IPv4 and IPv6-IPv6! IPv4-IPv6 we get a proper connection, secured with SRTP, but no audio. Looking at the network, RTP packets go from the caller to the server, but nothing leaves the server and no RTP packets go from callee to server. Do you have any idea how to fix this? Cheers, Mark

3 6

(no subject)
by Abdelkader Allam 04 Dec '13

04 Dec '13

Hello Guys, I am using Kamailio 4.1.0. I am trying to get Kamailio to rotate the cdr log through command kamcmd flatstore.rotate but it keeps sending log to the renamed files, I use the following parameter #!define WITH_ACCFILE # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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 */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) #!ifdef WITH_ACCFILE modparam("acc", "db_url", "flatstore:/var/log/kamailio") modparam("db_flatstore", "flush", 1) modparam("acc", "cdr_enable", 1) #modparam("acc", "cdr_flag", 3) modparam("acc", "db_extra","ds=$DLG_lifetime") modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$avp(i:256)") #!endif

1 0

Re: [SR-Users] COR
by Keith 04 Dec '13

04 Dec '13

Hi, I want to build in a Class of Restriction in Kamailio so I can allow/deny calls to numbers (countries) based on the caller CLI. Ideally I want to do this in the db but not sure the best way or if there is something to do this already? Cheers. Keith

2 1

"new_t: out_of_mem" errors, yet core.shmmem reports >60% shmmem free
by elactrum＠jamailca.com 04 Dec '13

04 Dec '13

Hi, I have a Kamailio 3.3.3 server that currently reports the following errors upon execution of t_relay() for certain initial INVITEs. daemon.err /usr/sbin/kamailio[26916]: ERROR: <core> [sip_msg_clone.c:505]: ERROR: sip_msg_cloner: cannot allocate memory daemon.err /usr/sbin/kamailio[26916]: ERROR: tm [t_lookup.c:1338]: ERROR: new_t: out of mem: daemon.err /usr/sbin/kamailio[26917]: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed I understand this error to be related to shared memory, based on [1]. However, sercmd reports normal shared memory utilization: # sercmd core.shmmem { total: 67108864 free: 43189728 used: 20505784 real_used: 23919136 max_used: 65948568 fragments: 36887 } These figures are quite similar to another server with the same configuration and Kamailio version, which is not reporting out-of-memory errors. # sercmd core.shmmem { total: 67108864 free: 40219536 used: 23241952 real_used: 26889328 max_used: 66306288 fragments: 37832 } I recently saw similar symptoms on a Kamailio 4.0.4 server, but Kamailio unfortunately crashed before I could get further details. In this case, I can consistently duplicate the problem with certain INVITEs. It seems that the failure is somehow related to the size of the SIP headers. For example, I have an INVITE from a Polycom phone which t_relay() routes successfully with 741 bytes of headers (total SIP length 1039 bytes). However, it seems that if I add one character to any of its SIP headers (742 bytes of headers; total SIP length 1039 bytes) the t_relay() will consistently fail. This binary doesn't have DBG_QM_MALLOC configured (I am working on getting it re-compiled). However, since I don't know what caused the problem, I would still like to get as much information as I can out of the running system. I have attached the output of mi_statistics, as well as mem_dump_pkg for the external UDP receivers and the ctl handler. Are there any other diagnostics that could be useful? -Andrew [1] http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:faq#qwhat_does_er…

3 3

Trouble getting Siremis up and running
by Mark D. Montgomery II 04 Dec '13

04 Dec '13

I just did a fresh from git install of kamailio and a fresh install of Siremis. Everything seemed to go fine until Siremis gets to the initial login. When I try to login I get a popup window labeled Debug Window with the following contents: System Internal Error The detailed error message are below: [2013-12-01 00:14:08 (GMT)] An exception occurred while executing this script: Error message: #256, Cannot find the class with name as service. Script name and line number of error: /var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php:157 function: errorHandler ( 256, "Cannot find the class with name as service.", "/var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php", 157, Array(12) ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/sysheader.inc 117 function: userErrorHandler ( 256, "Cannot find the class with name as service.", "/var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php", 157, Array(12) ) @ function: trigger_error ( "Cannot find the class with name as service.", 256 ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php 157 function: constructObject ( "service.securityService" ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php 56 function: getObject ( "service.securityService", 0 ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/BizSystem.php 252 function: getObject ( "service.securityService", 0 ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/BizSystem.php 241 function: getService ( "service.securityService" ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/BizController.php 57 function: processSecurityFilters ( ) @ /var/www/admin/siremis-4.0.0/openbiz/bin/BizController.php 31 function: include_once ( "/var/www/admin/siremis-4.0.0/openbiz/bin/BizController.php" ) @ /var/www/admin/siremis-4.0.0/siremis/bin/controller.php 6 Please ask system administrator for help... Report to admin Show Error Please wait a while ... ----------------- Any suggestions on what might be causing this? I'm not sure where to start looking. Thanks. Mark II -- Mark D. Montgomery II http://www.techiem2.net

2 1

How to build Kamailio without FUTEX reference (to use PTHREAD_MUTEX or POSIX_SEM instead)
by Sotas Development 04 Dec '13

04 Dec '13

Hi, I want to build a Kamailio version without the usage of futex. First of all I notice that the variable "use_futex" within Makefile.defs is hardcoded to the value "yes". Why is that? I want to get rid of this and build Kamailio without futex support. In the same file there are GCC options like -DUSE_PTHREAD_MUTEX, -DUSE_POSIX_SEM etc. So I disabled the line use_futex and and added -DUSE_PTHREAD_MUTEX as follows; #use_futex= yes C_DEFS+=-DUSE_PTHREAD_MUTEX After building Kamailio with these options: Although the output of "kamailio -I" indicates that FUTEX is not used (PTHREAD_MUTEX is used instead) it still uses futex sneaky as its locking mechanism. I can see this after catting the content of /proc/<pid>/wchan. This is because Kamailio hangs on futex after running it a while (see our other issue in the topic "Kamailio stability/timing problem w.r.t. registrations?"). Question is; How can I build Kamailio with a locking mechanism other than FUTEX? And why does the solution with Makefile.defs not work? PS: Kernel version is 2.6.30 on an ARM platform Thanks for your support. Kind regards, Orhan Yilmaz

2 1

Re: [SR-Users] [sr-dev] deb specs for 4.1
by Victor Seva 03 Dec '13

03 Dec '13

Hi there, this is my initial effort to get a new version [0]. This will create 4.2.0~dev0 packages for wheezy and jessie at master branch. I've just removed the older releases. I will maintain only this two. If everything is OK I will cherry-pick and create 4.1.0 packages at the 4.1 branch [0] https://github.com/linuxmaniac/kamailio/tree/vseva/deb PD: I've tried to create a personal branch without any luck. Is there any naming restriction on the git repo for that?

2 1

SIREMIS: Configure SIP Trunk/DID/Extension
by Stretchem 03 Dec '13

03 Dec '13

REPOST: Have successfully installed the latest stable Kamailio and Siremis. Have perused countless online Wiki's and other documentation, but can't seem to find any definitive tutorials or answers to the following: 1. Where would I set up a SIP registration to my carrier? (Server Services -> UACReg List?) 2. Where or how would I see the result of that registration? 3. Where would I set up DID's from the above carrier? 4. How would I inbound route those DID's to specific SIP extensions? (Have successfully registered extensions under Subscriber Services -> Subscriber List) 5. How would I outbound route calls from extensions to the carrier in #1? As I understand it, Kamailio settings are stored in a database; Siremis is a web gui to that database. If that is true, then I don't mind knowing the accompanying 'kamctl' commands to accomplish the above, since I'll see the respective settings in the Siremis web gui anyways. Also don't mind paying for the above tutorial, for which I'll turn into a Googlable blog entry. I need a routine or process for which non-CLI associates can make entries/edits. SIP Trunk with DIDs <--> Siremis on Kamailio <--> SIP Extension Thanks in advance.

1 0

deb specs for 4.1
by Daniel-Constantin Mierla 03 Dec '13

03 Dec '13

Initially sent to sr-dev, but maybe some people from users community can help as well... Hello, wondering if any debian/ubuntu fan has some time today/tomorrow morning for updating the specs to include the new modules coming with 4.1. The list is here: - http://www.kamailio.org/wiki/features/new-in-4.1.x Fortunately Peter updated the specs he maintains for rpm-based distros - the one for centos 6 might be useful to figure out dependencies - see in source tree pkg/kamailio/centos/6/. Reply quickly here if you start working on them -- there can be many people, I just want to know how I plan my effort to prepare the release. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Hello everyone.
by Yurisbel Jimenez Bermudez 03 Dec '13

03 Dec '13

I'm trying to setup Kamailio everything install correctly but I can't connect to my server with the users created. Not even in my local network it seems like the machine doesn't respond to the packets or maybe the packets are not delivery to my server but when I ping the machine everything is good I'm redirecting the port 5060 to my server same as range ports from 10000 to 20000 but can't connect Any one knows which is the problem here? I'm using Kamailio 4.0 on a virtual machine with virtual box. A copy of the same machine its running a web server of course with a different mac and IP and everything works fine.

2 1

Help required regarding error in Installation
by Owais ul Haq 03 Dec '13

03 Dec '13

Hello all, Have a nice day. I am given a task in office to setup Kamailio-3.1.1 on Fedora 14 machine. Being new to this, I went online for guidance. I got a proper method on the link given below and followed the instructions given. http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git But problem occured when I reached on step 4 i.e "COMPILE KAMAILIO". When I run command "MAKE ALL". Error occurs that asks for file "pcre.h" as shown in photo attached. Can some one please guide me what to do now ? Any help will be appreciated. Thanks. Regards, Kamailio newbie, Owais.

2 2

401 Unauthorized
by Joli Martinez 02 Dec '13

02 Dec '13

Hello all, I am trying to register a sip client to Kamailio. I have added the username password to kamctl and verified it in the opeser db that the user and password are correct. I have other users registered but I am unable to add new users. I keep getting the message below from tcpdump. Also kamctl online is not accurate. I have two other users connected and they dont show up there. [bad udp cksum a8b7!] UDP, length 433 E.......@.......2..........hSIP/2.0 401 Unauthorized Please help Thanks,

2 2

New developer: Luis Martin Gil
by Daniel-Constantin Mierla 02 Dec '13

02 Dec '13

Hello, I want to announce that a new person got developer GIT write access to repository: Luis Martin Gil. He has developed a new module named ndb_cassandra, which offers a connector to cassandra no-sql system directly from configuration style (e.g., for getting/setting items). The module is already part of the master branch since few days ago. His git commit id is: luismartingil 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

1 0

Next Kamailio World - April 2-4, 2014, in Berlin, Germany
by Daniel-Constantin Mierla 02 Dec '13

02 Dec '13

Hello, I would like to quickly announce the dates and place for next Kamailio World Conference: April 2-4, 2014, in Berlin, Germany. More information is available on the web site: - http://conference.kamailio.com/k01/2013/12/kamailio-world-2014-april-2-4-in… More details will be published in the near future. Looking forward to meeting many of you at the event in Berlin next year, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

forwarding doesn't work
by 1717 02 Dec '13

02 Dec '13

> > Greetings > > 1. Calling sip: mail(a)idd1717.com doesn't ring the forwarded sip uri, any idea? > 2. Can an sip alias name be enabled to be used for sip registration, just like the original user name? > > Thanks > James Wong

2 1

SIREMIS: Configure SIP Trunk/DID/Extension
by Stretchem 30 Nov '13

30 Nov '13

Have successfully installed the latest stable Kamailio and Siremis. Have perused countless online Wiki's and other documentation, but can't seem to find any definitive tutorials or answers to the following: 1. Where would I set up a SIP registration to my carrier? (Server Services -> UACReg List?) 2. Where or how would I see the result of that registration? 3. Where would I set up DID's from the above carrier? 4. How would I inbound route those DID's to specific SIP extensions? (Have successfully registered extensions under Subscriber Services -> Subscriber List) 5. How would I outbound route calls from extensions to the carrier in #1? As I understand it, Kamailio settings are stored in a database; Siremis is a web gui to that database. If that is true, then I don't mind knowing the accompanying 'kamctl' commands to accomplish the above, since I'll see the respective settings in the Siremis web gui anyways. Also don't mind paying for the above tutorial, for which I'll turn into a Googlable blog entry. I need a routine or process for which non-CLI associates can make entries/edits. SIP Trunk with DIDs <--> Siremis on Kamailio <--> SIP Extension Thanks in advance.

1 0

Kamailio for route traffic only
by driver 30 Nov '13

30 Nov '13

Hello, I have such infrastructure: Asterisk1 (provider 1) - IP1 Asterisk2 (provider 2) - IP2 Softphones on client computers with configured two IPs to two Asterisk servers. User has to manually switch between each server. I need to change this that user will have only one account configured (eg. Kamailio) and Kamailio will be used only to forward to Ast1 or Ast2 by previously defined rules. How is it possible? comp1 --------> Kamailio -------> Ast1 comp2 --------> Kamailio -------> Ast1 comp3 --------> Kamailio -------> Ast2 comp4 --------> Kamailio -------> Ast2 Thank you, Marcin

4 3

Some Help with Kamailio v4.0
by Tom Allen 30 Nov '13

30 Nov '13

I am spinning up my first instance of Kamailio and I can't seem to get it to start. I think my issue lies in the kamailio.cfg file, but I don't want to smash that entire file here, it is basically the default. I try to start and get this... root@kh1:/usr/local/etc/kamailio# /etc/init.d/kamailio debug start Starting Kamailio: loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: xxx:5060 udp: 192.168.xxxx:5060 tcp: 127.0.0.1:5060 tcp: xxxx:5060 tcp: 192.168.xxx:5060 Aliases: tcp: kh1:5060 tcp: kh1.xxxx.com:5060 tcp: localhost:5060 udp: kh1:5060 udp: kh1.xxxx.com:5060 udp: localhost:5060 kamailio error, failed to start. I am running a fresh load of Debian .. Linux kh1 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux I am using a mysql backend. Nothing is in production. What else can I provide you to help me figure this out? Thank you. Respectfully, Tom Allen Virtual Technologies Group

2 3

kamailio 4.0.3 - contact header in ACK
by Adnan 29 Nov '13

29 Nov '13

Hi, We are migrating from our openser 1.4 to kamailio 4.0.3 and have run into a couple of issues. One of them is described below and we need to know if it is a known bug or not. Kamailio receives an INVITE from a gateway and finally responds with 200 OK. Then an ACK is generated by the gateway towards the Kamailio that some how does not get relayed to the next hop. The only special aspect here is that the ACK does not contain a Contact header. Is that something that could confuse kamailio and make it unable to relay? According to SIP RFC, a contact header in ACK is not a must have. Our case is the one where ACK is sent by the UAC after receiving a 200 OK final response. Everything else about the ACK looks OK e.g. ACK has a branch different from the the branch value in the UAC INVITE. Is it a a known bug in 4.0.3? In the configuration, we handle ACK as below: if (is_method("ACK")) { if (t_check_trans()) { t_relay(); exit; } else { exit; } Br Adnan

3 4

Dialog module columns not found in .c files
by Grant Bagdasarian 29 Nov '13

29 Nov '13

Hello, Using 4.0.4: I'm getting the following error (removed the values part of the query): 3(2931) ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1. Query= insert into Dialogs (HashEntry,HashID,CallID,FromURI,FromTag,ToURI,ToTag,CallerSock,CalleeSock,StartTime,State,Timeout,CallerCSeq,CalleeCSeq,CallerRouteSet,CalleeRouteSet,CallerContact,CalleeContact,SFlags,ToRouteName,req_uri,xdata,iflags ) 3(2931) ERROR: db_unixodbc [con.c:220]: db_unixodbc_extract_error(): unixodbc:SQLExecDirect=42S22:1:207:[FreeTDS][SQL Server]Invalid column name 'req_uri'. The dialog.c file infact doesn't contain the req_uri_column, x_data_column and the iflags_column in static param_export_t mod_params[],static int mod_init(void). The dlg_db_handler.c files does contain them in the beginning of the file and in static int select_entire_dialog_table(db1_res_t ** res, int fetch_num_rows). I haven't checked the other remaining files, but I think the only problem is the renaming of these columns in the kamailio script. Just wondering if anyone else has noticed this?

1 0

what could be the problem here? 500 Invalid routeset?
by Kelvin Chua 29 Nov '13

29 Nov '13

U 2013/11/28 09:06:09.411796 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060 INVITE sip:+18X88X441X1@2X8.1X1.3X.2X SIP/2.0. Record-Route: <sip:6X.2X1.4.1X5;lr=on;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr=on;ftag=gK0662d671>. Accept: application/sdp. Allow: INVITE,ACK,CANCEL,BYE. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 INVITE. Max-Forwards: 47. Contact: <sip:+18X16X620X3@192.168.47.68:5060>. Content-Length: 327. Content-Disposition: session; handling=required. Content-Type: application/sdp. Remote-Party-ID: <sip:+18X16X620X3@192.168.47.68:5060 >;privacy=off;screen=no. . v=0. o=Sonus_UAC 30934 15833 IN IP4 192.168.47.68. s=SIP Media Capabilities. c=IN IP4 6X.2X1.4.9X. t=0 0. m=audio 5770 RTP/AVP 0 18 96 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:96 iLBC/8000. a=fmtp:96 mode=30. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:30. U 2013/11/28 09:06:09.413388 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 183 Session Progress. Record-Route: <sip:6X.2X1.4.1X5;lr=on;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr=on;ftag=gK0662d671>. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0;rport=5060. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X >;tag=bbd932f8f9dbf9743f9b7cadcbf622ac.10eb. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/11/28 09:06:09.413936 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0;rport=5060. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/11/28 09:06:10.672460 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 6X.2X1.4.1X5;rport=5060;branch=z9hG4bKcff3.da8920b2.0. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. Record-Route: <sip:2X8.1X1.3X.2X;lr;did=873.7d91;nat=yes>. Record-Route: <sip:6X.2X1.4.1X5;lr;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr;ftag=gK0662d671>. Call-ID: 1443253238_103264169(a)192.168.47.68. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. CSeq: 525 INVITE. Contact: <sip:1X5.6X.1X6.2X3:9439>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2013/11/28 09:06:12.931892 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 6X.2X1.4.1X5;rport=5060;branch=z9hG4bKcff3.da8920b2.0. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.0. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbcfc5fb404048790. Record-Route: <sip:2X8.1X1.3X.2X;lr;did=873.7d91;nat=yes>. Record-Route: <sip:6X.2X1.4.1X5;lr;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr;ftag=gK0662d671>. Call-ID: 1443253238_103264169(a)192.168.47.68. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. CSeq: 525 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:1X5.6X.1X6.2X3:9439>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length:238. . v=0. o=- 3594636366 3594636367 IN IP4 2X8.1X1.3X.2X. s=pjmedia. i=(o=IN IP4 10.0.193.243). c=IN IP4 1X5.6X.1X6.2X3. t=0 0. m=audio 10246 RTP/AVP 0. c=IN IP4 2X8.1X1.3X.2X. a=rtcp:10247. a=sendrecv. a=rtpmap:0 PCMU/8000. a=nortpproxy:yes. U 2013/11/28 09:06:12.960342 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060 ACK sip:1X5.6X.1X6.2X3:9439 SIP/2.0. Record-Route: <sip:6X.2X1.4.1X5;lr=on;ftag=gK0662d671>. Record-Route: <sip:6X.2X1.4.9X;lr=on;ftag=gK0662d671>. Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.2. Via: SIP/2.0/UDP 6X.2X1.4.9X;branch=z9hG4bKcff3.66c425c4.2. Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK06Bbd0344f60e48a125. From: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. To: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 525 ACK. Max-Forwards: 68. Route: <sip:2X8.1X1.3X.2X:5060;lr;did=873.7d91;nat=yes>. Content-Length: 0. . ------------------------ call goes on a for a while --------------------------- U 2013/11/28 09:06:24.358786 2X8.1X1.3X.2X:5060 -> 6X.2X1.4.1X5:5060 BYE sip:+18X16X620X3@6X.2X1.4.1X5:5060 SIP/2.0. Via: SIP/2.0/UDP 2X8.1X1.3X.2X;branch=z9hG4bKbb1b.267a7326.0. Via: SIP/2.0/UDP 1X5.6X.1X6.2X3:9439;rport=9439;branch=z9hG4bKPj6XR.AckX-G8lRIN4ZuDvjaPB6ZnUpHUa. Max-Forwards: 16. From: <sip:+18X88X441X1@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. To: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 5685 BYE. Route: <sip:6X.2X1.4.1X5;lr;ftag=gK0662d671>. Route: <sip:6X.2X1.4.9X;lr;ftag=gK0662d671>. User-Agent: CSipSimple_m0-16/r2330. Content-Length: 0. . U 2013/11/28 09:06:24.372741 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060 SIP/2.0 500 Invalid routeset. Via: SIP/2.0/UDP 2X8.1X1.3X.2X;branch=z9hG4bKbb1b.267a7326.0. Via: SIP/2.0/UDP 1X5.6X.1X6.2X3:9439;rport=9439;branch=z9hG4bKPj6XR.AckX-G8lRIN4ZuDvjaPB6ZnUpHUa. From: <sip:+18188244111@6X.2X1.4.9X>;tag=3pOAt1HRTOy9Xfi4Rsbbz9gf73MsuXge. To: <sip:+18X16X620X3@192.168.47.68;isup-oli=0>;tag=gK0662d671. Call-ID: 1443253238_103264169(a)192.168.47.68. CSeq: 5685 BYE. Server: Bandwidth.com CLEC. Content-Length: 0. . Kelvin Chua

4 15

Safe Maximum for Shared Memory?
by elactrum＠jamailca.com 28 Nov '13

28 Nov '13

Hi, I have Kamailio running on a 32-bit Linux server with 4 GB of memory. Is there any performance impact on Kamailio when increasing shared memory size? Or is the only concern that other processes have sufficient memory left over? The highest recommendation I have found on the mailing list was 1 GB [1], and one user reports using 2 GB [2]. Thanks! -Andrew [1] http://lists.sip-router.org/pipermail/sr-users/2011-October/070458.html [2] http://lists.sip-router.org/pipermail/users/2008-November/020646.html

2 1

rtpproxy / sdp / problem with signalling
by Oliver Roth 28 Nov '13

28 Nov '13

Problem with signalling - RTP gets lost! Rtpproxy not working properly? I am absolutely stuck ... cause this happens in a live environement. I have the following situation A calls B over carrier 1 - number is not valid and I get back error 404 from carrier and now freeswitch should play a message saying: "number not valid". But from carrier 1 I get back an RTP stream that is useless [1] - and if the correct streams opens from freeswitch - this does not get back to A [2]. I tested with rtpproxy on Kamailio - and all the rtp streams arrive at the Kamailio - but they cannot be "connected" correctly. I guess the problem is the 183 I get back from carrier 1 - after whitch rtp is opened. Or there is a wrong sdp singallisation if the "correct" stream arrives [3]. Sorry - I cannot get a solution - but I could provide various tcpdumps and pcaps. A Kamailio Carrier 1 Freeswitch INVITE ---------------------> 100 Your call is important <--------------------- INVITE ------------------------> 100 Trying <------------------------ 183 Session Progress SDP <------------------------ 183 Session Progress SDP <------------------------ RTP <================= RTP <================= RTP =================> RTP =================> [1] 404 not found <--------------------------- ACK ----------------------------> INVITE ---------------------------------------------------------> 100 Trying <-------------------------------------------------------- 200 OK SDP <-------------------------------------------------------- [3] 200 OK SDP <--------------------------- RTP (Announcment - number not valid") [2] <=================================== ACK ----------------------------> ACK ----------------------------------------------------------> INFO ----------------------------> INFO ----------------------------------------------------------> 200 OK <-------------------------------------------------------- 200 OK <--------------------------- BYE ----------------------------> BYE ----------------------------------------------------------> 200 OK <-------------------------------------------------------- 200 OK <---------------------------

2 1

Regarding Kamailio Sip Server
by Naveen Chand 28 Nov '13

28 Nov '13

Dear Sir, well i would like to know,is there any web interface for kamailio to configure complete configuration part,database,modules and upgrades similar like Astrix server ,Kamailio Sip server is Really Rock Solid server,when i first visited your site,i saw the name called Rock Solid ,i thought it just the call like that ,but after testing it ,it satisfied my needs,that is Video calling 2 way ,in excellent manner,well i would like to is Kamailio Sip server is Peer to Peer or not,if it is Peer to Peer,i have tested it ,i removed the connectivity from Server between Clients when call established,it worked for 20 mins and after it discards the call ,so i was bit confusing that kamailio sip server is Peer to Peer or not,pls confirm me,if it is peer to Peer,pls send me the step by step instruction how to configure the Peering module ,But i request you if there is any Web interface Like Astrix it would be great to control all the elements in Kamailio Best Regards T.NAVEEN CHAND

2 1

Books
by Joli Martinez 27 Nov '13

27 Nov '13

Hello, Is there any books you would recommend reading so I can learn more on Kamailio? thanks,

2 2

Log files
by Joli Martinez 27 Nov '13

27 Nov '13

I am running Kamailio in CentOS. I ran tcpdump and noticed that we are getting attacked from IP 188.138.32.72. I have already blocked it on IPtables, but he keeps on attacking the server. If I look at "/var/log/secure" there are no SIP messages. My question is where is the log file for Kamailio and how can I prevent this type of attacks in the future. Thanks,

5 10

voice prompts / early media and kamailio
by Oliver Roth 27 Nov '13

27 Nov '13

Hi all Based on my problem reported with subject "error handling<http://sip-router.1086192.n5.nabble.com/error-handling-tp123240.html>" I have some other questions. I think it is a conceptual question - and I do not see any solution. I would like to handle Kamailio with carrierroute / carrierfailureroute module different errors. Like 404, 403, busy - or whatever sip error occurs. Some of them need to be sent to a freeswitch playing an announcement (like "this number is blocked", "no more credit", ...). So if an error occur (lets say 403) then the call is routed by carrierfailurroute to fresswitch playing message for 403. If I am listening the whole message - I get back error 403 at the end and the call is logged in missed calls as error 403 sent from the freeswitch - everything ok. If I cancel the listening by hanging up - 487 is stored in missed_calls - cause I terminated the call before getting error 403 back from freeswitch. ==> so I loose this important information How can I get back error 403 - play an announcement and make sure, it is logged as 403 in database? Thanks for helping ... Oli

3 2

Re: [SR-Users] Problems with Siremis Register Page
by kamailio＠AaronLux.com 26 Nov '13

26 Nov '13

I am having the exact same problems as issue as this user. There is nothing in the siremis/log/ folder when this happens. I'm running Siremis 4.0.0 with Kamailio 4.0.4. Has this issue been fixed? Hello, On 5/18/13 3:22 AM, Alexander Albert wrote: > Hi there, i hope you can help me, i am having trouble with the > register page of siremis, strange thing is, i can open the register > page, but only after i have logged my self in, if not, it redirects me > directly to the login page.. how can i setup the register page to be > accessible without the need of logging in ? > Also i am getting the info after trying to register, that "Public > registration is not enabled!" > but i have set siremis/modules/ser/config/common.Main.php to > $cfg_siremis_public_registrations = true; > What is the problem, please help. can you look in siremis/log/ folder, there are some files printing debug messages, maybe you get hints about what is wrong. Cheers, Daniel

1 0

adding new column in the CDRS table?
by arun Jayaprakash 26 Nov '13

26 Nov '13

Hello, can someone let me know how I can add a new field in the cdrs table? I would like to set a flag depending if a call is make to another extension number of if the call is made to a PSTN line. Thank you for the help. Arun

2 2

Loosing rtp only in carrierroute failureroute
by Oliver Roth 26 Nov '13

26 Nov '13

Hi all We do have a strange problem with loosing rtp in case of carrierroute - failureroute. If we send traffic directly to the failure gateway, we do have rtp without any problem Situation: Ua --> Freeswitch --> kamailio 3.3 --> gw 1 Error 404 --> failureroute --> media gateway voiceprompt for error 404 ==> this does not work Ua --> Freeswitch --> kamailio 3.3 rewrite to dest for voiceprompt 404 -->media gateway voiceprompt for error 404 ==> this works!! Any idea? For me it seems being a problem with branching or so. Regards, Oli Configs below: Main routing block request_route { --- do all the checks route(CARRIERROUTE); route(RELAY); } route[RELAY] { xlog(, "L_INFO", "RELAY: Outbound sent via $avp(s:trunk_out)"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); xlog(,"L_INFO", "RELAX - Manage branch ...\n"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # carrierroute route[CARRIERROUTE] { xlog(,"L_INFO","Carrierroute module start \n"); $avp(s:tree) = "default"; # lookup from user and from domain cr_user_carrier("$fU", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)"); if($avp(s:cr_pref_carr) == 0){ # lookup from domain cr_user_carrier("", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)"); } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route 403 no route found $avp(s:cr_pref_carr)"); sl_send_reply("403", "Not allowed"); exit; } $avp(s:trunk_out) = $avp(s:todirection); route(ALTERHEADER); } failure_route[MANAGE_FAILURE] { if (t_grep_status("486")){ xlog("L_INFO", "Status 486 - busy"); t_reply("486", "Busy"); } #revert_uri(); route(NATMANAGE); if (t_is_canceled()) { exit; } xlog("L_INFO", "failure_route $rd $T_reply_code $avp(s:trunk_in), $avp(s:tree), $rU, $rd, $T_reply_code, $avp(s:tree) \n"); if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){ xlog("cr_next_domain failed"); exit; } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route failed"); exit; } $avp(s:trunk_out) = $avp(s:todirection); if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(FLT_ACCMISSED); # oro 28.10.13 } route(ALTERHEADER); t_on_failure("MANAGE_FAILURE"); if (!t_relay()) { xlog("failureroute t_relay failed"); exit; } }

1 1

route all calls to a wav file
by Joli Martinez 26 Nov '13

26 Nov '13

Hello, I am getting a lot of hits on port 5060. I would like all registered users to be sent to the correct location (currently working), but all unwanted users allow them to connect and any number they dial it will alway play a wav file. Is this possible if so how would I implement this. Thanks

3 2

Use of t_cancel_callid() to drop an early dialog
by Guillaume Bour 26 Nov '13

26 Nov '13

Hi All We wan't to prevent our users to make more than one call at time, so we choose to disconnect the previous call. When the previous call is established, we use dlg_bye(), and its ok. But when it is in early state, we use t_cancel_callid() to cancel its INVITE transaction. We face 2 issues: 1) we use local-request event route to account calls on timeout. Sometimes this route is called for the cancelled call (after default timeout of 1 hour) 2) t_cancel_callid() cancel previous call, but also _make current dialog disappear_: call is still ongoing and we can answer and talk to each other, but the dialog does not appear in 'kamctl stats dialog' and 'kamctl mi dlg_list' commands Is there a known limitation, or do we misuse t_cancel_callid() ? Regards, Guillaume

3 10

Setting up Conference Calls in Kamailio
by Mark Zeman 26 Nov '13

26 Nov '13

Hello everyone, We're thinking of setting up conference call capability for our Kamailio server, but we can't find any tutorials or documentation. Could someone point us in the right direction or maybe even tell us how to do it? Thanks!

2 1

Re: [SR-Users] Call hold and call conference implementation in kamailio
by Olle E. Johansson 26 Nov '13

26 Nov '13

26 nov 2013 kl. 14:22 skrev "Surendra" <surendra.pullaiah(a)plintron.com>: > Hi, > > Can we implement Call hold and Call conference features in kamailio. If how we can, else what will be the way to implement. > Can anyone help on this. These are media features better handled by end points and media servers. Kamailio by default doesn't handle media, it's a signalling server. Call HOld is between phones, so that's no issue. Call conference can be handled by using Asterisk, FreeSwitch or SylkServer and directing the calls that way in your Kamailio routing script. /O

1 0

Calls via Kamailio - Audio delay
by Mark Zeman 26 Nov '13

26 Nov '13

Hello everybody, having set up our Kamailio server and test env with Jitsi and CSipSimple successfully, we've noticed a delay of ~5 seconds in the audio. We are using IPv6 and encrypting everything (control and data channel) but this should still not lead to such a big delay, as far as we understand. Does anyone know why there would be this delay? Attached is our config, if that is of any help in fixing this delay. Cheers, Mark Zeman

2 1

Call hold and call conference implementation in kamailio
by Surendra 26 Nov '13

26 Nov '13

Hi, Can we implement Call hold and Call conference features in kamailio. If how we can, else what will be the way to implement. Can anyone help on this. Thanks in advance @surendra

1 0

ACL based on User-Agent
by Igor Potjevlesch 26 Nov '13

26 Nov '13

Hello, I was using an old unofficial module called 'check_ua.so' (with an old SER version) to check if an User-Agent is allowed or not. I don't find something similar to do this job in Kamailio. Maybe someone has already something workable with the existing modules? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 1

Problem with loose_route for a BYE
by Igor Potjevlesch 25 Nov '13

25 Nov '13

Hello, I work on interconnecting an old SER version with Kamailio. When a BYE is sent by the old SER, the "if (loose_route)" instruction returns false and then a "404 Not here" is sent. So the BYE is not relay to the UA. If the BYE is sent by Kamailio, no problem. Here are the BYE headers received by Kamailio: Record-Route: sip:OLD_SER_IP_ADDRESS;ftag=e97b7b77c5a3a9cc;lr=on Via: SIP/2.0/UDP OLD_SER_IP_ADDRESS;branch=z9hG4bKc9e5.3cbad207.0 Via: SIP/2.0/UDP UA_IP_ADDRESS_CONNECTED_TO_OLD_SER:5060;rport=5060;branch=z9hG4bK1fbe0a49fe3 44e6af1efd45b74305a9a Route: sip:KAMAILIO_IP_ADDRESS;lr=on I looked at the RR module documentation and it's clearly said that "The loose_route function analyzes the Route: headers in the requests. [.]If a Route: header is found, the function returns 1 and behaves as described in section 16.12 of RFC 3261.". Someone has an idea? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

3 6

error handling
by Oliver Roth 24 Nov '13

24 Nov '13

Hi all Question about error handling with kamailio. We send call to carrier and get back error 404. In carrierfailureroute we catch up this error and send call to an internal freeswitch that plays a voiceprompt saying: "destination not available" In accounting this calls is collected like a "normal" call - cause the internal freeswitch did the connection. I would like to play the voiceprompt but get the error 404 and see the call in the missed calls acc table. With 486 (busy) it is simple because we do not need an rtp response ... just fast busy. How can we handle this for errors we need to play a voiceprompt? What we do in failure route: failure_route[MANAGE_FAILURE] { sip_trace(); setflag(22); if (t_grep_status("486")){ xlog("L_INFO", "Status 486 - busy"); t_reply("486", "Busy"); exit; } revert_uri(); route(NATMANAGE); if (t_is_canceled()) { exit; } xlog("L_INFO", "failure_route rd: $rd replCode: $T_reply_code trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU: $rU, todirection $avp(s:todirection) fu: $fu\n"); if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){ xlog("cr_next_domain failed"); exit; } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route failed"); exit; } xlog("L_INFO", "failure_route rd: $rd replCode: $T_reply_code trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU: $rU, todirection $avp(s:todirection) fu: $fu \n"); $avp(s:trunk_out) = $avp(s:todirection); xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via $avp(s:trunk_out) rU $rU -------------------------------------------------------- "); if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(FLT_ACCMISSED); # oro 28.10.13 } route(ALTERHEADER); t_on_failure("MANAGE_FAILURE"); #append_branch(); if (!t_relay()) { xlog("failureroute t_relay failed"); exit; } }

3 5

Re: [SR-Users] SEG FAULT dialog
by Kelvin Chua 24 Nov '13

24 Nov '13

I can reproduce it consistently. I am doing serial forking into a non-existent host, when the caller gets impatient, it cancels the call and send another invite. this happens after that. full backtrace: #0 qm_detach_free (frag=0x7fb2b552e2b0, qm=<optimized out>) at mem/q_malloc.c:269 prev = 0x7fb2b5192058 next = 0x0 #1 qm_malloc (qm=0x7fb2b5192000, size=16) at mem/q_malloc.c:386 f = 0x7fb2b552e2b0 hash = -1256644520 #2 0x00007fb2b44f7e93 in set_dlg_variable_unsafe () from /usr/local/lib64/kamailio/modules/dialog.so No symbol table info available. #3 0x00007fb2b44fa3aa in pv_set_dlg_variable () from /usr/local/lib64/kamailio/modules/dialog.so No symbol table info available. #4 0x000000000047be59 in lval_pvar_assign (rv=0x7fb2bf8348f8, lv=0x7fb2bf834750, msg=0x7fb2bf8c7a00, h=<optimized out>) at lvalue.c:353 r_avp = <optimized out> ret = 1 destroy_pval = 1 pvar = 0x7fb2bf834758 pval = {rs = { s = 0x92750b "+1xx882xx111(a)2xx.xx1.3x.2xSIP/2.0\r\nRecord-Route: <sip:6x.x1.4.195;lr=on;ftag=gK085a1dec>\r\nRecord-Route: <sip:6X.2X1.8.8X;lr=on;ftag=gK085a1dec>\r\nAccept: application/sdp\r\nAllow: INVITE,ACK,CANCEL,BYE"..., len = 12}, ri = 0, flags = 4} avp_val = {n = 0, s = {s = 0x0, len = 0}, re = 0x0} v = <optimized out> #5 lval_assign (h=<optimized out>, msg=0x7fb2bf8c7a00, lv=0x7fb2bf834750, rve=0x7fb2bf8348f0) at lvalue.c:401 rv = 0x7fb2bf8348f8 ret = 0 __FUNCTION__ = "lval_assign" #6 0x000000000041cf14 in do_action (h=0x7fffe1b8f710, a=0x7fb2bf8332b0, msg=0x7fb2bf8c7a00) at action.c:1453 ret = -5 v = <optimized out> dst = {send_sock = 0x7fb2bdb0f080, to = {s = {sa_family = 0, sa_data = "\000\000\000\000\000\000\200-\210\000\000\000\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\200-\210\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "\200-\210\000\000\000\000\000\016\000\000\000\000\000\000", __u6_addr16 = {11648, 136, 0, 0, 14, 0, 0, 0}, __u6_addr32 = {8924544, 0, 14, 0}}}, sin6_scope_id = 13}}, id = 0, proto = 58 ':', send_flags = {f = 180 '\264', blst_imask = 138 '\212'}} tmp = <optimized out> new_uri = <optimized out> end = <optimized out> crt = <optimized out> cmd = <optimized out> len = <optimized out> user = <optimized out> uri = {user = {s = 0x7fffe1b8eef0 "/", len = 0}, passwd = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507973920}, host = { s = 0x7fffe1b8ee20 "", len = 17022112}, port = {s = 0x7fb2bffbd720 "", len = 8208}, params = { s = 0x2010 <Address 0x2010 out of bounds>, len = 17030320}, sip_params = { s = 0x2010 <Address 0x2010 out of bounds>, len = -1077387804}, headers = { s = 0x7fb2bffbdfb0 "0z\001\001", len = -507974144}, port_no = 55072, proto = 49147, type = 32690, flags = 17022128, transport = {s = 0x95 <Address 0x95 out of bounds>, len = -1241200896}, ttl = { s = 0x3734000000a0 <Address 0x3734000000a0 out of bounds>, len = -1081931184}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x63371f "", len = -1}, method = {s = 0x7fb2b519cb30 "\220\240H\265\262\177", len = 4310469}, lr = {s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507973440}, r2 = { s = 0x7fffe1b8f000 "\210u\223", len = 8187}, gr = {s = 0x2 <Address 0x2 out of bounds>, len = 1}, transport_val = {s = 0x1 <Address 0x1 out of bounds>, len = 0}, ttl_val = { s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, len = 17022112}, user_param_val = { s = 0x7fb2bffbd720 "", len = -507974064}, maddr_val = {s = 0x7fb2bf8c7a00 "\016", len = -1081931184}, method_val = {s = 0x2010 <Address 0x2010 out of bounds>, len = -507971824}, lr_val = {s = 0x0, len = -507971824}, r2_val = {s = 0x7fffe1b8f580 "\001", len = -1}, gr_val = { s = 0x7fb2b519cb30 "\220\240H\265\262\177", len = 4346308}} next_hop = {user = {s = 0x0, len = 1}, passwd = {s = 0x7fffe1b8f710 "\002", len = 1237187532}, host = { s = 0x7fffe1b8f710 "\002", len = -507972224}, port = { s = 0xffffffff <Address 0xffffffff out of bounds>, len = -1256600784}, params = { s = 0x3542f9ae92fdfbcc <Address 0x3542f9ae92fdfbcc out of bounds>, len = -589431860}, sip_params = { s = 0x0, len = 17022128}, headers = {s = 0x15 <Address 0x15 out of bounds>, len = 1}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 4294967200, transport = {s = 0x7fffe1b8efe8 "l\345N", len = -1076936721}, ttl = {s = 0x1 <Address 0x1 out of bounds>, len = -1114716768}, user_param = { s = 0x1c0000002f <Address 0x1c0000002f out of bounds>, len = 7}, maddr = { s = 0x710000000a <Address 0x710000000a out of bounds>, len = 2}, method = {s = 0x0, len = -18000}, lr = {s = 0x1017a30 "EST", len = 6323844}, r2 = {s = 0x7fb2bd8ed356 "tm [t_lookup.c:716]: ", len = -507972224}, gr = {s = 0x7fffe1b8f710 "\002", len = 5148368}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x7fb2bf8c7a00 "\016", len = -507973348}, user_param_val = { s = 0x103bcb0 "<135>Nov 19 07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 0}, maddr_val = { s = 0x103bcb0 "<135>Nov 19 07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = -1115163744}, method_val = { s = 0x99 <Address 0x99 out of bounds>, len = 1384864127}, lr_val = {s = 0x882d80 "", len = -1081312768}, r2_val = {s = 0x7fb2bdb0f080 "\016", len = 8924544}, gr_val = { s = 0xffffffff <Address 0xffffffff out of bounds>, len = -1081930768}} u = <optimized out> port = <optimized out> dst_host = <optimized out> i = <optimized out> flags = <optimized out> avp = <optimized out> st = {flags = 3213654528, id = 32690, name = {n = -507971824, s = {s = 0x7fffe1b8f710 "\002", len = -507972224}, re = 0x7fffe1b8f710}, avp = 0x4ee56c} sct = <optimized out> sjt = <optimized out> rve = <optimized out> mct = <optimized out> rv = <optimized out> rv1 = <optimized out> c1 = {cache_type = 3213020352, val_type = 32690, c = {avp_val = {n = 0, s = { s = 0x7fb200000000 <Address 0x7fb200000000 out of bounds>, len = -1081312768}, re = 0x7fb200000000}, pval = {rs = {s = 0x7fb200000000 <Address 0x7fb200000000 out of bounds>, len = -1081312768}, ri = -1124131338, flags = 32690}}, i2s = "\377\377\377\377\060", '\000' <repeats 16 times>} s = {s = 0x7fffe1b8f0d8 "\260\062\203\277\262\177", len = -507973632} srevp = {0x937588, 0x7fb2bfc4f1fc} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = { s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #7 0x000000000041c5c5 in run_actions (h=0x7fffe1b8f710, a=0x7fb2bf8313a0, msg=0x7fb2bf8c7a00) at action.c:1573 t = <optimized out> ret = <optimized out> mod = <optimized out> ms = 0 __FUNCTION__ = "run_actions" #8 0x000000000041d92e in do_action (h=0x7fffe1b8f710, a=0x7fb2bf8396d0, msg=0x7fb2bf8c7a00) at action.c:1065 ret = 1 v = 1 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = "\000\000\001\000\000\000\000\000\000\000\000\000\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 1}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = { sin6_family = 0, sin6_port = 0, sin6_flowinfo = 1, sin6_addr = {__in6_u = { __u6_addr8 = "\000\000\000\000\000\000\000\000\001\000\000\000\000\000\000", __u6_addr16 = {0, 0, 0, 0, 1, 0, 0, 0}, __u6_addr32 = {0, 0, 1, 0}}}, sin6_scope_id = 0}}, id = 0, proto = -80 '\260', send_flags = {f = 188 '\274', blst_imask = 3 '\003'}} tmp = <optimized out> new_uri = <optimized out> end = <optimized out> crt = <optimized out> cmd = <optimized out> len = <optimized out> user = <optimized out> uri = {user = {s = 0x3635353531000002 <Address 0x3635353531000002 out of bounds>, len = 1}, passwd = { s = 0x0, len = 17022128}, host = {s = 0x7fffe1b8f460 "/", len = 0}, port = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507972528}, params = { s = 0x7fffe1b8f390 "\260\337\373\277\262\177", len = -1076936721}, sip_params = { s = 0xa2 <Address 0xa2 out of bounds>, len = -1114690816}, headers = { s = 0x1c0000002f <Address 0x1c0000002f out of bounds>, len = 7}, port_no = 10, proto = 0, type = 113, flags = URI_SIP_USER_PHONE, transport = {s = 0x0, len = -18000}, ttl = {s = 0x1017a30 "EST", len = -1076935806}, user_param = {s = 0x7fb2b5531d90 "P\213\065\265\262\177", len = 48}, maddr = { s = 0x3734e1b8f3b8 <Address 0x3734e1b8f3b8 out of bounds>, len = -507972896}, method = { s = 0x3531320000000002 <Address 0x3531320000000002 out of bounds>, len = -1076363582}, lr = { s = 0x103bcbc "07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 17022140}, r2 = {s = 0x7fb2bffbb3c0 "", len = 7}, gr = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507972048}, transport_val = { s = 0x7fffe1b8f570 "\001", len = 8187}, ttl_val = {s = 0x2 <Address 0x2 out of bounds>, len = 1}, user_param_val = {s = 0x1 <Address 0x1 out of bounds>, len = 0}, maddr_val = { s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, len = 17022112}, method_val = { s = 0x7fb2bffbd720 "", len = 8208}, lr_val = {s = 0x2010 <Address 0x2010 out of bounds>, len = 17030320}, r2_val = {s = 0x2010 <Address 0x2010 out of bounds>, len = -1077387804}, gr_val = { s = 0x7fb2bffbdfb0 "0z\001\001", len = -1077164088}} next_hop = {user = {s = 0x7fb2bffbd720 "", len = 8180}, passwd = { s = 0x3635353531000002 <Address 0x3635353531000002 out of bounds>, len = 17022112}, host = {s = 0x0, len = -1077381378}, port = {s = 0x7fffe1b8f5f0 "", len = 0}, params = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507972128}, sip_params = { s = 0x7fffe1b8f520 "\002", len = -1077164088}, headers = {s = 0x0, len = 8180}, port_no = 2, proto = 12544, type = 909456693, flags = 17022128, transport = {s = 0x0, len = 1}, ttl = { s = 0x7fffe1b8f650 "0\313\031\265\262\177", len = 0}, user_param = { s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -507972032}, maddr = { s = 0x7fffe1b8f580 "\001", len = 6526864}, method = { s = 0x37340000002f <Address 0x37340000002f out of bounds>, len = 7}, lr = { s = 0x320000710000000a <Address 0x320000710000000a out of bounds>, len = -1076363582}, r2 = { s = 0x103bcbc "07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 17022140}, gr = {s = 0x7fb2bffbb3c0 "", len = 7}, transport_val = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507971648}, ttl_val = { s = 0x7fffe1b8f700 "0\313\031\265\262\177", len = 8187}, user_param_val = { s = 0x373400000002 <Address 0x373400000002 out of bounds>, len = 1}, maddr_val = { s = 0x1 <Address 0x1 out of bounds>, len = -1076363582}, method_val = { s = 0x103bcbc "07:28:47 /usr/local/sbin/kamailio[15556]: DEBUG: dialog [dlg_hash.c:602]: dlg_lookup(): dialog id=7094 found on entry 2049\n", len = 17022140}, lr_val = {s = 0x7fb2bffbb3c0 "", len = 7}, r2_val = { s = 0x3000000018 <Address 0x3000000018 out of bounds>, len = -507971552}, gr_val = { s = 0x7fffe1b8f760 "", len = 8187}} u = <optimized out> port = <optimized out> dst_host = <optimized out> i = <optimized out> flags = <optimized out> avp = <optimized out> st = {flags = 2, id = 0, name = {n = 1, s = {s = 0x1 <Address 0x1 out of bounds>, len = 1}, re = 0x1}, avp = 0x0} sct = <optimized out> sjt = <optimized out> rve = <optimized out> mct = <optimized out> rv = <optimized out> rv1 = <optimized out> c1 = {cache_type = 137, val_type = RV_NONE, c = {avp_val = {n = 17022112, s = {s = 0x103bca0 "`\002", len = 160}, re = 0x103bca0}, pval = {rs = {s = 0x103bca0 "`\002", len = 160}, ri = -1077381378, flags = 32690}}, i2s = " \327\373\277\262\177\000\000\000\000\000\000\001\000\000\000\340\366\270\341\377\177"} s = {s = 0xffffffffffffffa0 <Address 0xffffffffffffffa0 out of bounds>, len = 17022112} srevp = {0x7fb2bffbd720, 0x2010} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = { s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #9 0x000000000041c5c5 in run_actions (h=0x7fffe1b8f710, a=0x7fb2bf82d4c0, msg=0x7fb2bf8c7a00) at action.c:1573 t = <optimized out> ret = <optimized out> mod = <optimized out> ms = 0#0 qm_detach_free (frag=0x7fb2b552e2b0, qm=<optimized out>) at mem/q_malloc.c:269 prev = 0x7fb2b5192058 next = 0x0 ... cut .... Kelvin Chua On Tue, Nov 19, 2013 at 9:40 PM, Daniel-Constantin Mierla <miconda(a)gmail.com > wrote: > Hello, > > can you get the output for 'bt full'? Probably you have to install the > debug symbols (kamailio-dbg package) for getting something useful. > > Another question, can you reproduce it? Or it happens sporadically? > > Cheers, > Daniel > > > > > On 11/19/13 1:11 PM, Kelvin Chua wrote: > > kamailio 4.0.4 > has anybody experience this using dialog module? > > backtrace: > #0 0x000000000057af45 in qm_malloc () > #1 0x00007f268ac0fe93 in set_dlg_variable_unsafe () from > /usr/local/lib64/kamailio/modules/dialog.so > #2 0x00007f268ac123aa in pv_set_dlg_variable () from > /usr/local/lib64/kamailio/modules/dialog.so > #3 0x000000000047be59 in lval_assign () > #4 0x000000000041cf14 in do_action () > #5 0x000000000041c5c5 in run_actions () > #6 0x000000000041d92e in do_action () > #7 0x000000000041c5c5 in run_actions () > #8 0x0000000000425250 in run_top_route () > #9 0x00000000004bfcce in receive_msg () > #10 0x000000000056a2df in udp_rcv_loop () > #11 0x0000000000481b43 in main_loop () > #12 0x000000000041c081 in main () > > logfile: > DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: > DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> > DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: > <sip:+1XX882XX111@2XX.XX1.X9.2X> > DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> > DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param > type 232, <branch> = <z9hG4bKe97b.5a4397f3.0>; state=16 > DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header > reached, state=5 > DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: > Via found, flags=2 > DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: > this is the first via > DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... > DEBUG: <core> [receive.c:192]: receive_msg(): preparing to run routing > scripts... > DEBUG: <core> [parser/parse_to.c:176]: parse_to_param(): DEBUG: add_param: > tag=bbd932f8f9dbf9743f9b7cadcbf622ac.0dc7 > DEBUG: <core> [parser/parse_to.c:799]: parse_to(): end of header reached, > state=29 > DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: > get_hdr_field: <To> [74]; uri=[sip:+1XX882XX111@6X.X31.X.8X] > DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [ > <sip:+1XX882XX111@6X.X31.X.8X>] > DEBUG: sl [sl_funcs.c:415]: sl_filter_ACK(): SL local ACK found -> > dropping it! > DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: > destroying list (nil) > last message repeated 5 times > DEBUG: <core> [xavp.c:447]: xavp_destroy_list(): destroying xavp list (nil) > DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up > segfault at 0 ip 000000000057af45 sp 00007fff30e44ac0 error 4 in > kamailio[400000+27c000] > > Kelvin Chua > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Trainings - Berlin, Nov 25-28 > - more details about Kamailio trainings at http://www.asipto.com - > > > _______________________________________________ > 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 4

kamailio 4.0.4 package for ubuntu
by Freddy Andersen 22 Nov '13

22 Nov '13

I’m currently running Kamailio 4.0.3 in production and think I’m hitting bugs that looks like they are fixed in 4.0.4 but deb.kamailio.org does not have 4.0.4 yet? Any timeline when this will get released? -- - Freddy

1 0

ubuntu package for kamailio 4.0.4
by Freddy Andersen 22 Nov '13

22 Nov '13

I’m currently running Kamailio 4.0.3 in production and think I’m hitting bugs that looks like they are fixed in 4.0.4 but deb.kamailio.org does not have 4.0.4 yet? Any timeline when this will get released? -- - Freddy

1 0

Kamailio returns "405 Method Not Allowed" for SIP MESSAGE request
by Karthikeyan R 22 Nov '13

22 Nov '13

Is Kamailio supports SIP MESSAGE request? Please let me know how to configure. Regards, Karthikeyan R DISCLAIMER: This email message and all attachments are confidential and may contain information that is Privileged, Confidential or exempt from disclosure under applicable law. If you are not the intended recipient, you are notified that any dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please notify us immediately by return email to mailadmin(a)spanservices.com and destroy the original message. Opinions, conclusions and other information in this message that do not relate to the official of SPAN, shall be understood to be nether given nor endorsed by SPAN.

3 2

siptrace & sipcapture - write to multiple tables
by Grant Bagdasarian 22 Nov '13

22 Nov '13

Hello, Is it possible to have the sipcapture module write the duplicated messages to multiple tables? If not, is it possible to have the siptrace module duplicate the message to multiple Sip Capture instances? Regards, Grant

3 3

add carrier
by Joli Martinez 21 Nov '13

21 Nov '13

Hello, I have Kamailio installed as an SBC with Freeswitch. I would like to add a SIP trunk to Kamailio. Is there any documentation on how to do this? Also, is it recommended to add trunks to Kamailio vs Freeswitch if so why? Thanks

3 5

REGISTER request's "Via" address used to send INVITE
by Karthikeyan R 21 Nov '13

21 Nov '13

Hi All, I'm trying to configure the Kamailio Server (V 4.0.3) as Registrar with OverSIP server (V 1.4.1) as a Proxy for my WebRTC client; While calling UserB from UserA, the Kamailio is trying to find the UserB using address from Via address in UserB's REGISTER request; I'm also using GRUU and enabled the same in configuration file. Can someone help me what's wrong in my configuration / is Kamailio requires any additional settings to resolve this issue? Attached Files.zip contains the following files: OverSip-Kamailio.pcapng - Contains the packet capture between the OverSIP and Kamailio Servers. Kamailio-Logs.txt - Shows the error message that Kamailio is trying to resolve address and it's failed. User-records.txt - Output of "kamctl ul show". Kamailio.cfg - Configuration file for Kamailio. Note: I have configured MySQL as DBENGINE, but haven't created any DB tables. Thanks in Advance, Karthikeyan R DISCLAIMER: This email message and all attachments are confidential and may contain information that is Privileged, Confidential or exempt from disclosure under applicable law. If you are not the intended recipient, you are notified that any dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please notify us immediately by return email to mailadmin(a)spanservices.com and destroy the original message. Opinions, conclusions and other information in this message that do not relate to the official of SPAN, shall be understood to be nether given nor endorsed by SPAN.

1 0

Problem with 4.0 nightlies and IMS - sem_post undefined
by Barry Flanagan 21 Nov '13

21 Nov '13

Hi I am trying to run the Ubuntu nightlies from http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf config files. Things were working fine while I was on the 20130110 nightlies, but when I upgraded to 20130122 I get the following errors. Any have an idea why sem_post is undefined and how to fix this? 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib64/kamailio/modules/ims_registrar_pcscf.so>: /usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib64/kamailio/modules/cdp.so>: /usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib64/kamailio/modules/ims_qos.so>: /usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post Thanks -Barry Flanagan

4 10

append_hf to CANCEL
by Grant Bagdasarian 21 Nov '13

21 Nov '13

Hello, I'm executing the following code for both the BYE and CANCEL in request_route: if (is_method("BYE|CANCEL")) { #finished $var(finished) = $TS; append_hf("X-Finished: $var(finished)\r\n"); } For some reason, the X-Finished header is appended to the BYE, but not the CANCEL. The CANCEL is forwarded further. I have the setflag(22); set, so all outbound messages are sent to the Capture server. Does append_hf not work for CANCEL requests?

4 9

Transaction Matching Not Working
by Samuel Ware 21 Nov '13

21 Nov '13

I am currently working on building a type of outbound proxy. I am having an issue with t_check_trans() not finding the associated transaction. I thought maybe this was because I was manipulating the RURI and other headers so I turned on the called_matching parameter and turned off ruri_matching and via1_matching. I get the same result as before making that change of the t_check_trans() not returning a positive value. # Check source address/port of request if ($si != "xxx.xxx.xxx.xxx") { # This means that the server communicating is not an internal # sip signaling server. We must determine in the message is # part of an existing Call Dialog. If not, we drop the message. # If so, relay the message on presumably to the appropriate internal # server. if ( t_check_trans() ) { # Confirmed part of an existing dialog, so we relay xlog("L_INFO","External Server ($si:$sp) - Sent Relay Message (Call ID: $ci METHOD: $rm)"); route(RELAY); exit; } else { # Message is not part of an existing dialog, so drop it with # no response xlog("L_WARN","External Server ($si:$sp) - Sent Non-Authorized Message (Call ID: $ci METHOD: $rm)"); exit; } }; SIP Capture: U 2013/11/20 17:08:16.87172 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 INVITE sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:mod_sofia@xxx.xxx.xxx.xxx:5060> User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130723T192601Z~1dbcfbfa02 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 205 X-VSIP-CUST-IP: 162.243.64.6 X-AUTH-IP: 162.243.64.6 X-VSIP-OUTBOUND: DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060 X-FS-Support: update_display,send_info P-Asserted-Identity: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx> v=0 o=FreeSWITCH 390420642 390420643 IN IP4 xxx.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 19532 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 U 2013/11/20 17:08:16.87340 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport=5060;branch=z9hG4bKceQ2e5g58evvD From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Server: kamailio (4.0.4 (x86_64/linux)) Content-Length: 0 U 2013/11/20 17:08:16.87387 yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:5060 INVITE sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060 SIP/2.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;branch=z9hG4bK9dcc.06fa6927.0 ;rport=5060Max-Forwards: 14 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:mod_sofia@xxx.xxx.xxx.xxx:5060> User-Agent: VSIP-OP v0.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 205 X-AUTH-IP: 162.243.64.6 X-FS-Support: update_display,send_info P-Asserted-Identity: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx> v=0 o=FreeSWITCH 390420642 390420643 IN IP4 xxx.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 19532 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 U 2013/11/20 17:08:16.117314 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Server: MERA MVTS3G v.4.3.0-46 Content-Length: 0 U 2013/11/20 17:08:18.383674 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 183 Progress Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Server: MERA MVTS3G v.4.3.0-46 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:18.570976 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:19.71773 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:20.72864 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:22.74578 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:26.73588 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:30.72173 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:32.187227 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:32.686940 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:33.688358 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:34.71527 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:35.688296 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:38.70512 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:39.691038 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:42.74073 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:43.689345 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:46.77664 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:47.689095 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:50.76714 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK9dcc.06fa6927.0 Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 INVITE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Content-Type: application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE Server: MERA MVTS3G v.4.3.0-46 X-mera-expires: 86460 Content-Length: 243 v=0 o=- 1384967471 1384967471 IN IP4 209.165.2.198 s=- c=IN IP4 209.165.2.198 t=0 0 m=audio 32278 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - U 2013/11/20 17:08:51.688846 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:55.689470 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:08:55.903509 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:08:56.904308 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:08:58.903685 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:08:59.689703 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:09:02.903777 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:03.690873 zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060 BYE sip:mod_sofia@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP zzz.zzz.zzz.zzz:5060;rport;branch=z9hG4bK-2095418569-3809543762-352366480-125028071 Via: SIP/2.0/UDP 10.11.12.224:5061;rport=5061;branch=z9hG4bK-2095418569-3809543762-352366480-12502807;received=10.11.12.224 Route: <sip:yyy.yyy.yyy.yyy;lr=on> From: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy>;tag=3234265279-3809543762-352365712-12502807 To: <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147671 BYE Contact: <sip:DST_PHONE_NUMBER@zzz.zzz.zzz.zzz:5060> Max-Forwards: 70 User-Agent: MERA MVTS3G v.4.3.0-46 Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 U 2013/11/20 17:09:06.902640 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:10.904529 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:14.903813 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:18.906135 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:22.902976 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 U 2013/11/20 17:09:26.905600 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060 CANCEL sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;rport;branch=z9hG4bKceQ2e5g58evvD Max-Forwards: 15 From: "" <sip:SRC_PHONE_NUMBER@xxx.xxx.xxx.xxx>;tag=Ug3cBSK17a9Ng To: <sip:DST_PHONE_NUMBER@yyy.yyy.yyy.yyy> Call-ID: 97537438-cca9-1231-a897-0c010ab38601 CSeq: 52147670 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0

3 2

WebRTC - XOR-MAPPED-ADDRESS and bridge mode
by Kelvin Chua 20 Nov '13

20 Nov '13

here is the scenario: kamailio 4.0.4 running in bridge mode eth0 : 10.17.0.202 eth1 : 100.200.30.40 a call comes in from ice enabled webrtc client (chrome) from public internet, kamailio processed the call, no audio. if i replace kamailio with asterisk, audio ok. upon investigating using wireshark, i found out that chrome sends a "Binding Request" for the IP/port in ICE candidate. Kamailio/rtpproxy sends back Binding Success XOR-MAPPED-ADDRESS but the IP is the internal one 10.17.0.202:port this is not right, the response should be the remote IP/port, not the local IP/port Kelvin Chua

1 1

Does $hdr() accept PVs?
by Alex Balashov 20 Nov '13

20 Nov '13

That is to say, can I do? $var(x) = 'P-Asserted-Identity'; xlog("L_INFO", "PAI: $hdr(var(x))\n"); What about remove_hf("$var(x)"); -- Alex -- 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

EOF Error in pass_fd
by Ricardo Martinez 20 Nov '13

20 Nov '13

Hello List. The last weekend our kamailio process crashed with this error : Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10661]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 29 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: <core> [main.c:751]: child process 10601 exited by a signal 11 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: <core> [main.c:754]: core was generated Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD Can someone tell me what seems to be the problem here? We’re using : version: kamailio 3.2.2 (x86_64/linux) 98ba92-dirty 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, DBG_QM_MALLOC, USE_FUTEX, 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, epoll_lt, epoll_et, sigio_rt, select. id: 98ba92 -dirty compiled on 12:45:36 Feb 2 2012 with gcc 4.4.6 And this is the bt full from gdb Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 mi_profile_list (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_profile.c:789 789 if ( mi_print_dlg( rpl, ph->dlg, 0)!=0 ) 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 full #0 mi_profile_list (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_profile.c:789 node = <value optimized out> rpl_tree = 0x20a6790 rpl = 0x20a67b0 profile = 0x7fe99a2dda00 ph = 0x0 profile_name = <value optimized out> value = <value optimized out> i = <value optimized out> #1 0x00007fe9dce287b1 in run_mi_cmd (fifo_stream=<value optimized out>) at ../../lib/kmi/mi.h:77 No locals. #2 mi_fifo_server (fifo_stream=<value optimized out>) at fifo_fnc.c:509 mi_cmd = 0x208f8f0 mi_rpl = <value optimized out> hdl = 0x0 line_len = 41 file_sep = 0x2039017 "" command = 0x7fe9dfdbdef9 "profile_list_dlgs" file = 0x7fe9dfd98d78 "/tmp/openser_receiver_27809" f = 0x2039018 reply_stream = 0x20b0980 #3 0x00007fe9dce2a640 in fifo_process (rank=<value optimized out>) at mi_fifo.c:247 fifo_stream = 0x209a0b0 #4 0x00007fe9dce2a9de in mi_child_init (rank=0) at mi_fifo.c:211 pid = <value optimized out> #5 0x00000000004dba41 in init_mod_child (m=0x7fe9dfc6ba10, rank=0) at sr_module.c:886 No locals. #6 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6bd08, rank=0) at sr_module.c:883 No locals. #7 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6c248, rank=0) at sr_module.c:883 No locals. #8 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6c748, rank=0) at sr_module.c:883 No locals. #9 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6d288, rank=0) at sr_module.c:883 No locals. #10 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6d928, rank=0) at sr_module.c:883 No locals. #11 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6dee0, rank=0) at sr_module.c:883 No locals. #12 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e1f0, rank=0) at sr_module.c:883 No locals. #13 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e5d0, rank=0) at sr_module.c:883 No locals. #14 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e830, rank=0) at sr_module.c:883 No locals. #15 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6ebd0, rank=0) at sr_module.c:883 No locals. #16 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6f220, rank=0) at sr_module.c:883 No locals. #17 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6f648, rank=0) at sr_module.c:883 No locals. #18 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6fa60, rank=0) at sr_module.c:883 No locals. #19 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6fe40, rank=0) at sr_module.c:883 No locals. #20 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc702c0, rank=0) at sr_module.c:883 No locals. #21 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc70ae8, rank=0) at sr_module.c:883 No locals. ---Type <return> to continue, or q <return> to quit--- #22 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc70e18, rank=0) at sr_module.c:883 No locals. #23 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc71100, rank=0) at sr_module.c:883 No locals. #24 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc716c0, rank=0) at sr_module.c:883 No locals. #25 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc71df0, rank=0) at sr_module.c:883 No locals. #26 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc723c0, rank=0) at sr_module.c:883 No locals. #27 0x000000000046077d in main_loop () at main.c:1661 i = <value optimized out> pid = <value optimized out> si = 0x0 si_desc = "udp receiver child=15 sock=64.76.154.35:5060", '\000' <repeats 20 times>, "\003", '\000' <repeats 15 times>, "\030\325\a\232\351\177\000\000\001\000\000\000\002", '\000' <repeats 11 times>, "\001\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000)\000\000\000\000\000\000" #28 0x0000000000461f13 in main (argc=<value optimized out>, argv=0x7fff2f2b2848) at main.c:2475 cfg_stream = <value optimized out> c = <value optimized out> r = <value optimized out> tmp = 0x7fff2f2b3f76 "" tmp_len = 0 options = 0x5a3178 ":f:cm:M:dVhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1850310455 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = <value optimized out> n_lst = <value optimized out> p = <value optimized out> (gdb) Regards, Ricardo.-

4 7

change callerid, kamailio
by vladimir tarasov 20 Nov '13

20 Nov '13

Substituted by callerid mobile number - the higher the operator beats, and other provider passes. How to write so that the directions to change the phone number that is displayed with customers from 7452393 to 3493357505

1 0

change callerid
by vladimir tarasov 20 Nov '13

20 Nov '13

How do I keep calling with an asterisk on the kamailio, changed callerid on kamailio and then called on the gw?

1 0

remove a line from SDP
by Laurent Schweizer 20 Nov '13

20 Nov '13

Dear all, I try to remove a line from an SDP with: subst_body('/^a=fmtp:8 vad=no(.*)$//g'); unfortunately the "0a" of the end of line ( 0d0a ) is not removed, so the "$" remove only the 0d any idea how to remove the last "0a" Best regards Laurent

2 2

TLS questions
by Fabian Borot 20 Nov '13

20 Nov '13

Hi, please help me out with these questions: Is it possible to configure kamailio to use different certificates, one per peer? Do we just add them to the same "calist.pem" file? Is is possible to configure kamailio to accept a certificate from another device? I mean instead of Kamailio creating the certificate/key and give it to customer/provider. If that is the case, do we also add it to the "calist.pem" file as well? If all the above is true, what happens when the certificates have different expiration dates? thank you very much

3 7

kamailio and db_cassandra
by jay binks 20 Nov '13

20 Nov '13

Hi all, can anyone comment on the lack of a debian package for the db_cassandra module ?? I understand there may be reasons to not distribute the binary, but I was really hoping to find a control file so I can build my own package. any hints / pointers on the best way to achive this and the reasons behind it would be great. -- Sincerely Jay

3 7

need help regarding with kamailio3.3
by Manav Kumar 20 Nov '13

20 Nov '13

Hi , I need some help regarding kamailio 3.3 presence issue. Below I mentioned the scenario which I'm doing with kamailio3.3. I simply install kamailio3.3 with configuration file(as per http://kb.asipto.com/kamailio:presence:k31-made-simple site) which I have attached with this mail. I have started my kamailio on tcp as well as UDP because I want getting presence with xcap configuration and after configured jitsi when I open the jitsi client it makes the tcp connection with kamailio server and fetch users list from server by following request ( http://ip:port/ xcap-root/resource-lists/users/sip:manav@172.21.183.138/index) and jitsi client gives pop up message i.e http://ip:port/xcap-root/xcap-caps/global/index resource cannot be read . and when I see in wireshark traces it closes the tcp connection . Can you help me this issue as I need some kind of resolution on high priority ? Thanx in advance for your support. Regards, Manav =============================================================================== Please refer to http://www.aricent.com/legal/email_disclaimer.html for important disclosures regarding this electronic communication. ===============================================================================

1 0

record_route_preset() Question
by kamailio＠AaronLux.com 20 Nov '13

20 Nov '13

Why did 'transport=tls' 'r2=on' 'lr=on' and 'nat=yes' all disappear after just adding 'record_route_preset("myurl.com");' (I had to edit my routing logic to fix a problem with kamailio sending record-route to WAN clients telling them to respond on my private NAT IP address) # I'm concerned my edits to the routing logic is incorrect and would like a second opinion. (I added record_route_preset on line 41 and commented out line 42): http://pastebin.com/3azFStmE # Here is what record-route looked like Before adding record_route_preset (3 lines): Record-Route: <sip:66.111.222.111:5061;transport=tls;r2=on;lr=on;nat=yes> Record-Route: <sip:66.111.222.111;r2=on;lr=on;nat=yes> Record-Route: <sip:myurl.com;lr=on> # And now here is what they look like After adding record_route_preset: (1 line) Record-Route: <sip:myurl.com;lr> MISC INFO: # kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 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, USE_FUTEX, 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, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7

1 0

SEG FAULT dialog
by Kelvin Chua 19 Nov '13

19 Nov '13

kamailio 4.0.4 has anybody experience this using dialog module? backtrace: #0 0x000000000057af45 in qm_malloc () #1 0x00007f268ac0fe93 in set_dlg_variable_unsafe () from /usr/local/lib64/kamailio/modules/dialog.so #2 0x00007f268ac123aa in pv_set_dlg_variable () from /usr/local/lib64/kamailio/modules/dialog.so #3 0x000000000047be59 in lval_assign () #4 0x000000000041cf14 in do_action () #5 0x000000000041c5c5 in run_actions () #6 0x000000000041d92e in do_action () #7 0x000000000041c5c5 in run_actions () #8 0x0000000000425250 in run_top_route () #9 0x00000000004bfcce in receive_msg () #10 0x000000000056a2df in udp_rcv_loop () #11 0x0000000000481b43 in main_loop () #12 0x000000000041c081 in main () logfile: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:+1XX882XX111@2XX.XX1.X9.2X> DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKe97b.5a4397f3.0>; state=16 DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... DEBUG: <core> [receive.c:192]: receive_msg(): preparing to run routing scripts... DEBUG: <core> [parser/parse_to.c:176]: parse_to_param(): DEBUG: add_param: tag=bbd932f8f9dbf9743f9b7cadcbf622ac.0dc7 DEBUG: <core> [parser/parse_to.c:799]: parse_to(): end of header reached, state=29 DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [74]; uri=[sip:+1XX882XX111@6X.X31.X.8X] DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:+1XX882XX111@6X.X31.X.8X>] DEBUG: sl [sl_funcs.c:415]: sl_filter_ACK(): SL local ACK found -> dropping it! DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) last message repeated 5 times DEBUG: <core> [xavp.c:447]: xavp_destroy_list(): destroying xavp list (nil) DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up segfault at 0 ip 000000000057af45 sp 00007fff30e44ac0 error 4 in kamailio[400000+27c000] Kelvin Chua

2 1

kamdbctl create ?
by Thomas Secula 18 Nov '13

18 Nov '13

I seem to be in a catch 22, sure it's something I've done wrong. I set dbhost in kamctlrc to be the ip of my remote mysql server I run kamdbctl create and I get root@pcscf:/etc/kamailio# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... ERROR 1044 (42000) at line 1: Access denied for user 'root'@'172.16.101.27' to database 'kamailio' ERROR: granting privileges to database kamailio failed! root@pcscf:/etc/kamailio# Before I run the create from mysql: mysql> select user,host from mysql.user; +------+---------------+ | user | host | +------+---------------+ | root | % | | root | 172.16.101.27 | | root | 172.16.101.28 | | root | 172.16.101.29 | | root | localhost | +------+---------------+ 5 rows in set (0.00 sec) mysql> show grants for 'root'@'%'; +--------------------------------------------------------------------------- -----------------------------------+ | Grants for root@% | +--------------------------------------------------------------------------- -----------------------------------+ | GRANT ALL PRIVILEGES ON *.* TO 'root'@'%' IDENTIFIED BY PASSWORD '*FE04940F3F9B339AB9361B2D01AD3D940B215B52' | +--------------------------------------------------------------------------- -----------------------------------+ 1 row in set (0.00 sec) If I rerun it the create scripts says: root@pcscf:/etc/kamailio# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... ERROR 1007 (HY000) at line 1: Can't create database 'kamailio'; database exists ERROR: Creating database kamailio failed! root@pcscf:/etc/kamailio# any ideas?

2 3

kamailio tls.reload core dump
by Ding Ma 18 Nov '13

18 Nov '13

Hi, all This is related to the previous tls.reload not safe email chain. Now we have a detailed gdb output that shows the stack trace of the core dump. Please take a look. This looks like a bug. Please let me know if you have any insights on how to fix this. Thanks, Ding

2 6

kamailio 4.0 crash at urecord wb_timer
by jh＠tutpro.com 18 Nov '13

18 Nov '13

this kind of crash has occurred a few times now with kamailio 4.0: Program terminated with signal 11, Segmentation fault. #0 0xb56b41bd in wb_timer (_r=<optimized out>) at urecord.c:319 319 urecord.c: No such file or directory. (gdb) where #0 0xb56b41bd in wb_timer (_r=<optimized out>) at urecord.c:319 #1 0xb56a9602 in mem_timer_udomain (_d=0xa59574e8, istart=0, istep=1) at udomain.c:879 #2 0xb569ae3e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:742 #3 0xb56b0fff in destroy () at ul_mod.c:489 #4 0x0811408c in destroy_modules () at sr_module.c:790 #5 0x0809c6ec in cleanup (show_status=1) at main.c:573 #6 0x0809d529 in shutdown_children (show_status=1, sig=<optimized out>) at main.c:715 #7 0x0809e800 in handle_sigs () at main.c:806 #8 0x0809fe18 in main_loop () at main.c:1767 #9 0x080a1bbe in main (argc=17, argv=0xbff02f74) at main.c:2566 would be nice to get it fixed. -- juha

2 6

Siremis Getting Started - error when logging in
by Mark D. Montgomery II 17 Nov '13

17 Nov '13

I got Siremis installed today but when I try to login it pops up a debug window with the error: Fatal error: Cannot find the class with name as service. in /var/www/admin/siremis-4.0.0/openbiz/bin/ObjectFactory.php on line 157 Any suggestions on what I need to check to fix this so I can actually login and start using it? Thanks. Mark II -- Mark D. Montgomery II http://www.techiem2.net

1 0

Timezone in acc table
by Igor Potjevlesch 16 Nov '13

16 Nov '13

Hi, I'm wondering how can I change the timezone when INVITE, ACK, BYE and CANCEL are inserted into acc table? Indeed, today, each entrie has the same datetime than the server. I would prefer to have UTC time in "time" column instead. I know that it was the behaviour with old SER releases. Thank for your help. Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

3 3

Fwd: sr-users Digest, Vol 102, Issue 53
by santosh satapathy 15 Nov '13

15 Nov '13

---------- Forwarded message ---------- From: santosh satapathy <santosh.satapathy(a)gmail.com> Date: Fri, Nov 15, 2013 at 4:23 PM Subject: Fwd: sr-users Digest, Vol 102, Issue 53 To: Daniel-Constantin Mierla <miconda(a)gmail.com> Appreciate for the input Daniel. I was looking here little different scenario where in weights are dynamic. Load distribution should happen in such a way that first it round robins among the nodes with weight 3 until they are equal with other sets of nodes with weight 2. Once this stage is reached where in all the nodes in both the sets are equally weighted, then LB should round robin among all the nodes uniformly. This means that dynamically nodes with higher weights will get more calls until their weight difference is substantiated and after that all nodes are treated as equal. Is it possible to configure script for the above scenario? Thanks , Santosh ---------- Forwarded message ---------- From: <sr-users-request(a)lists.sip-router.org> Date: Thu, Nov 14, 2013 at 2:26 AM Subject: sr-users Digest, Vol 102, Issue 53 To: sr-users(a)lists.sip-router.org 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: Kamailio/Radius problem (Daniel-Constantin Mierla) 2. Re: Variables in event_route[sl:local-response] (Daniel-Constantin Mierla) 3. Re: Query regd the dispatcher algorithm support (Daniel-Constantin Mierla) 4. WORDS_BIGENDIAN error in auth_diameter.h (Sotas Development) ---------------------------------------------------------------------- Message: 1 Date: Thu, 14 Nov 2013 10:06:20 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio/Radius problem Message-ID: <5284928C.9080300(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" On 11/14/13 10:02 AM, Daniel-Constantin Mierla wrote: > Hello, > > On 11/14/13 4:31 AM, jonathan augenstine wrote: >> I am working on the configuration to enable RADIUS in Kamailio 4.0. >> The basic install worked successfully and the daemon started without >> problems. I then install radiusclient-ng and configured RADIUS in >> the Kamailio build/configuration. All seemed OK until the Kamailio >> restart and I encountered the following error during the restart: >> >> Nov 13 06:52:23 lax-vproxy2-a kernel: [ 348.808624] kamailio[5346]: >> segfault at 0 ip 00007f375d249a5d sp 00007fff4d42eed0 error 4 in >> libradiusclient-ng.so.2.0.0[7f375d243000+c000] >> Nov 13 06:52:23 lax-vproxy2-a kamailio: ERROR: <core> >> [daemonize.c:307]: daemonize(): Main process exited before writing to >> pipe >> >> After some research it seemed that others encountered a similar error >> and traced it to the dictionary. I have gone through the dictionary >> configuration and all seems to be correct. Any guidance on what >> might be the issue or some troubleshooting steps? > did you get a core dump? Looking at the backtrace may show where is > the problem in libradiusclient-ng. Just found this one that seems to be similar: - http://sip-router.org/tracker/index.php?do=details&task_id=310&project=1 It mentions that acctserver was not set in radius config. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Kamailio + freeswitch BLF is not working
by José Seabra 15 Nov '13

15 Nov '13

Hello there, Im trying implement BLF with kamailio and snom phones, with the following modules: presence, presence_xml, presence_mwi, pua_dialoginfo, presence_dialoginfo, pua and dialog calls between phones registed in kamailio, BLF works fine. The issue is, when the call comes from freeswitch --> Kamailio --> snom phone, Kamailio creates a Publish but doesn't send notify's to the phones that are monitoring the ringing phone. the logs shows the following: DEBUG: presence [notify.c:1249]: publ_notify(): Could not find subs_dialog Nov 15 16:36:47 vx00-kml01 /usr/local/sbin/kamailio[13953]: DEBUG: <core> [msg_translator.c:2026]: build_res_buf_from_sip_res(): build_res_from_sip_res: copied size: orig:666, new: 596, rest: 302 msg=#012SIP/2.0 180 Ringing#015#012Via: SIP/2.0/UDP 94.46.248.80;received=10.0.20.10;rport=5060;branch=z9hG4bKDgmN81t5NZrDg#015#012Record-Route: <sip:94.46.XX.XX:5060;r2=on;lr=on;ftag=ZvNgee3Uy81tr;lb=yes>#015#012Record-Route: <sip:10.0.20.7:5062;r2=on;lr=on;ftag=ZvNgee3Uy81tr;lb=yes>#015#012Record-Route: <sip:10.0.20.13;lr=on;ftag=ZvNgee3Uy81tr;proxy=yes;did=213.ae12;nat=yes>#015#012From: "256370XXX" <sip:25637XXX@94.46.2X.XXX>;tag=ZvNgee3Uy81tr#015#012To: < sip:+3513045XXXX@admin.com>;tag=fvt7h9az6c#015#012Call-ID: 3ef55a52-4e14-11e3-85cd-71b7082538d9#015#012CSeq: 51930666 INVITE#015#012Contact: <sip:203@192.168.1.180:32768;alias=10.0.20.7~5062~1;alias=213.228.167.67~32768~1;line=sg3y775p>;reg-id=1#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE#015#012Allow-Events: talk, hold, refer, call-info#015#012Content-Length: 0#015#012P-VOICIS-Src-Ip: 213.228.167.67#015#012P-VOICIS-Src-Port: 32768#015#012P-VOICIS-Src-Proto: udp#015#012#015#012 Nov 15 16:36:47 vx00-kml01 /usr/local/sbin/kamailio[13957]: DEBUG: presence [notify.c:1249]: publ_notify(): Could not find subs_dialog Nov 15 16:36:47 vx00-kml01 /usr/local/sbin/kamailio[13955]: DEBUG: presence [notify.c:1249]: publ_notify(): Could not find subs_dialog Anyone can help on this? -- Regards José Seabra

1 0

Presentation of Kamailio v4.1.0 today on VUC
by Daniel-Constantin Mierla 15 Nov '13

15 Nov '13

Hello, a reminder for those that want to participate later today to the VoIP Users Conferences presenting what is new in Kamilio's upcoming release v4.1.0 as well as discussing the future plans. For interactive participation, you can join: - for voice - dial in via sip, skype or pstn - see http://vuc.me - for text chat - join irc channel #vuc on server freenode.net For passive participation, you can: - watch the video streaming - listen the audio streaming The streaming links will be published on http://vuc.me when the session starts (based on past sessions, they should be http://vuc.me/live or http://live.vuc.me for video and http://vuc.me/listen or http://vuc.me/m3u for audio). The session is scheduled to start at 17:00GMT - to learn the time in your zone, click on http://vuc.me/next In the video session will be a consistent group of developers, more details are at: - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

roni@pomvan.co.uk
by Roni | 15 Nov '13

15 Nov '13

Hi I would like to please join the mailing list

2 1

Kamailio/Radius problem
by jonathan augenstine 15 Nov '13

15 Nov '13

I am working on the configuration to enable RADIUS in Kamailio 4.0. The basic install worked successfully and the daemon started without problems. I then install radiusclient-ng and configured RADIUS in the Kamailio build/configuration. All seemed OK until the Kamailio restart and I encountered the following error during the restart: Nov 13 06:52:23 lax-vproxy2-a kernel: [ 348.808624] kamailio[5346]: segfault at 0 ip 00007f375d249a5d sp 00007fff4d42eed0 error 4 in libradiusclient-ng.so.2.0.0[7f375d243000+c000] Nov 13 06:52:23 lax-vproxy2-a kamailio: ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited before writing to pipe After some research it seemed that others encountered a similar error and traced it to the dictionary. I have gone through the dictionary configuration and all seems to be correct. Any guidance on what might be the issue or some troubleshooting steps? Jonathan

2 4

Kamailio Segfault Dialog.so and http_query inside event_route
by Juan Luis Faylon 15 Nov '13

15 Nov '13

Hi, I need help. I encounter a segfault error in dialog.so when I do an http_query containing shared pseudovariables in the event_routes of dialog:start and dialog:end. How can I fix this?

2 1

kamailio sending qop in auth header
by Daniel W. Graham 14 Nov '13

14 Nov '13

It may be my lack of understanding, but I moved from using auth_check function to www_authenticate function and now see kamailio sending qop in auth header. It's my understanding this is not necessary (based on UA types in use) so I would like to remove it. Per module doc, it is "" by default, I tried adding auth param as below but made no difference. What could be causing this? U x.x.x.x:5060 -> y.y.y.y:5060 REGISTER sip:y.y.y.y SIP/2.0. Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-bd3965f2. From: user <sip:user@y.y.y.y>;tag=682bdce181bb953eo0. To: user <sip:user@y.y.y.y>. Call-ID: 1b886b6c-b283de31(a)x.x.x.x. CSeq: 612 REGISTER. Max-Forwards: 70. Contact: user <sip:100@x.x.x.x:5060>;expires=1800. User-Agent: agent Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. . U y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-bd3965f2;rport=5060. From: user <sip:user@y.y.y.y>;tag=682bdce181bb953eo0. To: user <sip:user@y.y.y.y>;tag=02500e456b34f8655aebe61627730a5a.852c. Call-ID: 1b886b6c-b283de31(a)x.x.x.x. CSeq: 612 REGISTER. WWW-Authenticate: Digest realm="y.y.y.y", nonce="UoV8q1KFe38amcNXXHXmuDU83y1OXg2n", qop="auth". Server: SIP Proxy. Content-Length: 0.. version: kamailio 4.0.3 (i386/linux) 090035 # ----- auth params ----- modparam("auth", "qop", "") # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "id") modparam("auth_db", "use_domain", MULTIDOMAIN) modparam("auth_db", "version_table", 0) ......... if (!www_authenticate("$td", SIP_TABLE)) { www_challenge("$td", "1"); exit; } ......... [cid:image001.gif@01CEE17B.A12FBB50]

1 1

(no subject)
by Roni | 14 Nov '13

14 Nov '13

may i join

4 3

GIT Branch 4.1 has been created for release series 4.1.x
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, the stable branch from where we are going to package the release series 4.1.x has been created - its name: 4.1 There is now a 'kamailio' alias for the repository name, therefore it can be used instead of 'sip-router' in the git path. Getting the git branch 4.1 with read only access: git clone --depth 1 --no-single-branch git://git.sip-router.org/kamailio kamailio cd kamailio git checkout -b 4.1 origin/4.1 For developers that have write access: git clone --depth 1 --no-single-branch ssh://_USERID_@git.sip-router.org/kamailio kamailio cd kamailio git checkout -b 4.1 origin/4.1 Note that older versions of git application may not have the option --no-single-branch. Just remove it when is the case. I created also the wiki tutorial about installing kamailio 4.1.x from git, it is available at: - http://www.kamailio.org/wiki/install/4.1.x/git After few more days of testing, we can loot at nailing down the release date for v4.1.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Development for v4.2.0 is open
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, a dedicated branch 4.1 has been created for upcoming major release. From now on, the master can take again code with new features to be part of future release series 4.2.x. Any new bug fix that is applicable to stable branches has to be committed first in master and then backported. I will follow with another email describing how to use branch 4.1 for those that want to switch testing from master to it. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Kamailio VUC Session on Nov 15
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, we are preparing for a new VUC session to give an update about Kamailio project - a perfect timing as we are just about to release a new major version. We will like to get many developers involved to be able to highlight properly what is new, especially those that did new development for v4.1 - new modules or enhancement to existing modules. However, any developer and community member is welcome to join, we will appreciate it very much, in support of the project. Those that have a google hangout account can participate with video, otherwise there are options to join via sip or pstn audio calls - you can see more details at: - http://vuc.me The number of participants with video is limited, therefore if you plan to do it, let me know to be able to coordinate and send you the invite link when the session starts. No matter you participate with audio/video, you can join the IRC channel #vuc on freenode.net for text chatting during the event. I made quickly a news about the event: - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/ I will add names of other participants as I get confirmations. Do suggest topics to highlight/discuss there as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

3 3

Disable NAT for given IP range
by David Dean 14 Nov '13

14 Nov '13

Hello all, I have a kamailio server on a public IP address with a private interconnect into my office network. The private interconnect allows the server to see requests from individual 10.x.x.x IP addresses instead of everything coming through the public NAT. This avoids timeout issues and reduces keepalive traffic. The problem is the SIP headers from clients on the 10.x.x.x IP range match nat_uac_test because of the RFC1918 address in the via and contact headers and so NAT support is enabled regardless of no NAT being present. Is it possible to disable NAT support for a given IP range, in my case 10.x.x.x ? Thanks, John

3 2

WORDS_BIGENDIAN error in auth_diameter.h
by Sotas Development 14 Nov '13

14 Nov '13

Hello All, We are currently cross compiling Kamailio for powerpc and are encountering an "issue", of which we want to know whether or not it is intentional. In the modules/auth_diameter/diameter_msg.h of Kamailio 4.0.4 (latest) we found the following line: 72: #error BIG endian detected!! But it looks like if all the code is in place to make big endian function as desired. Is this a leftover debug message? The reason we are asking is because we previously encountered another error with a similar check for WORDS_BIGENDIAN, in MD5.h. It appears as the check is built in on multiple places for this preprocessor directive, however it is not set anywhere. Are we missing a necessary build step somewhere? We have solved the issue for now by patching the diameter_msg.h to remove the #error statement, and start the makefile using: CC_EXTRA_OPTS = "-DWORDS_BIGENDIAN" This seems to solve our initial issue, which was an md5 mismatch on authentication, however we are still unsure if this is a proper workaround, or if we are missing something else. It would be very nice if someone could shed some light on this. Thanks in advance, Sotas Team (Pascal, Yuri).

1 0

Query regd the dispatcher algorithm support
by santosh satapathy 14 Nov '13

14 Nov '13

Does the dispatcher module allow combination like "weighted round robin" algorithm. Example: 3 of 5 nodes are in better condition so I set them as weight 1 and the other two as weight 2. Then the dispatcher will round robin between the 3 nodes with weight 1 until all 5 are with the same weight. That also means that the weight should be dynamic and updated as conditions of the node changes. -- Thanks Santosh

2 2

Variables in event_route[sl:local-response]
by Sergey Okhapkin 14 Nov '13

14 Nov '13

Which variables I can use in event_route[sl:local-response] route to xlog proto:ip:port to which the response will be sent? event_route[sl:local-response] { xlog("L_INFO","Local response to *$??*\n$mb\n"); }

3 2

Kamailio 4.0.4 crash t_load_contact();
by Julia 14 Nov '13

14 Nov '13

Hello, After upgrade from 3.3 to 4.0.4 Kamailio crashed when the first call after application start is a call with failure route: get_redirect(); t_load_contact(); t_next_contacts(); Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:78]: get_redirect(): resume branch=0 Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:85]: get_redirect(): checking branch=0 (added=0) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:89]: get_redirect(): branch=0 is a redirect (added=0) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:165]: sort_contacts(): sort_contacts: <sip:39988202@10.25.147.241:5060;user=phone> q=250 Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:165]: sort_contacts(): sort_contacts: <sip:39988202@10.25.148.41:5060;user=phone> q=500 Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:297]: shmcontact2dset(): adding contact <sip:39988202@10.25.147.241:5060;user=phone> Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: <core> [sruid.c:176]: sruid_next(): new sruid is [rdir-5280b7a9-2874-1] (1 / 20) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: uac_redirect [rd_funcs.c:297]: shmcontact2dset(): adding contact <sip:39988202@10.25.148.41:5060;user=phone> Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: <core> [sruid.c:176]: sruid_next(): new sruid is [rdir-5280b7a9-2874-2] (2 / 20) Nov 11 12:55:48 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10356]: DEBUG: tm [t_serial.c:166]: t_load_contacts(): nr_branches is 2 Nov 11 12:56:00 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10336]: ALERT: <core> [main.c:788]: handle_sigs(): child process 10356 exited by a signal 11 Nov 11 12:56:00 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10336]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated Nov 11 12:56:00 vm-KAMnet-dev01 /usr/local/sbin/kamailio[10336]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLD Regards, Julia.

2 6

Steps to next major release v4.1.0
by Daniel-Constantin Mierla 14 Nov '13

14 Nov '13

Hello, it's now one month since we froze the development for release of v4.1.0. No critical issues in on the table for this particular version, so perhaps next week is time to create a dedicated branch for it, to be named in GIT as 4.1, and open development for v4.2. If all goes fine for one more week or so, then we can do the release. Don't forget to add any open issues you are aware of to the tracker so we can solve them in time. If any of devs or community members have spare time, adding ids to the xml docbook files (for modules) will help getting a better alpha-numeric index for parameters and functions -- see more at: - http://www.kamailio.org/wiki/devel/module-docbook-readme#section_ids Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

3 6

Stuck dialogs
by Daniel W. Graham 13 Nov '13

13 Nov '13

I'm occasionally getting a stuck dialog in my database. I use dialog module for enforcing call limits. I found this post but still not sure what to do: http://lists.sip-router.org/pipermail/sr-users/2013-April/077681.html What should I look for / how should I troubleshoot this issue? -dan

2 8

forward audio to another sip server
by Joli Martinez 13 Nov '13

13 Nov '13

I have all user registration done through kamailio and conferences are handled by freeswitch. I would like to forward all audio to another sip server where the conference can be recorded and we can pull reports. I understand that FS can do the recording, but a box has been setup for this and this is the way my boss wants it. please let me know how to accomplish this thanks,

2 1

extract part of string / INVITE
by Oliver Roth 13 Nov '13

13 Nov '13

Hi all I need to extract a part of the INVITE msg: INVITE sip:+41796159999@82.197.185.185;user=CSC10824 SIP/2.0 I need 10824 in a avp variable. Is there a regex function where I can extract this? Or how can this be done? Something like: Avp(myCSC) = string after CSC - length 5 Please be aware, that ip and destination number in the invite can change ... Otherwise substring would be a solution. Thanks for help Oli

4 3

DBTEXT / AVPOPS and custom database
by Hugh James 13 Nov '13

13 Nov '13

Thanks Daniel, That helped my understanding further along , but it wasn't the full story . If I use $var(rc)=avp_db_load("$to/username","$avp(inbound)/LineIDs"); Then I still get no matches . But if I place entries in the uuid column of the table , then it works. I had assumed that since I don't use the uuid or the domain they could be null . In fact I tried it again with just a number in the uuid column of the entry I expected the avp_db_load to match , and it still worked . I then placed a single entry in the uuid column of a row that I didn't expect it to match and it still worked . I then placed two identical numbers in the UUID column and it still works ( see below for current table used in testing) So in summary it appears that there must be at least one value in the uuid column for it to work as expected . Cheers Hugh

1 0

Kamailio, Keepalived: Kamailio doesn't respond on virtual IP but real IP
by Nguyễn Đức Vinh 13 Nov '13

13 Nov '13

Hi all I want to setup IP failover for Kamailio: I have 2 hosts A (IP: X.X.X.32) and B (X.X.X.36), with 2 running kamailios. Using Keepalive to setup IP failover as , virtual IP *is X.X.X.43*. On host A, kamailio was binded to *listen=udp:0.0.0.0:5060 <http://0.0.0.0:5060>* and host A is MASTER with virtual IP *Problem is:* Client send REGISTER to X.X.X.43:5060. It recieved 401 respond from X.X.X.32:5060 (actual IP of A), not as expected from X.X.X.43:5060 (To check if keepalived setting OK or not, I have *nc -l 5060* on host A, and using nc X.X.X.43 5060 on client, wireshark show that responds is from X.X.X.43 -> keepalived work OK Is there any difference between *kamailio* and *nc* cause they both bind to 0.0.0.0, but repond on differ IP? Thanks Vinh Nguyen

4 3

Function to test DB connection in the cfg file?
by Allen Zhang 12 Nov '13

12 Nov '13

Hi all, This is a re-post. My email didn't seem to get on the list last time. Is there a function I can use to test database connection from the routing script in the cfg file? My dispatcher uses OPTIONS probing. When the destination receives the OPTIONS request, I'd like to detect the health status of the database connection before sending a reply back. Note the OPTIONS request doesn't have a user part in the RURI. Hence, is_subscriber() and alias_db_lookup() don't work. Any suggestion? Regards, Allen Zhang

5 5

Forwarding registrations upstream
by Spencer Thomason 12 Nov '13

12 Nov '13

Hello, I'm trying to use Kamailio as a customer facing SBC to handle NAT traversal and basic SBC functionality for a group of backend FreeSWITCH servers. What I'm trying to accomplish is this: - UAC sends REGISTER - Auth is handled by the edge proxy, backend server trusts edge proxy - If NAT is detected, expires is decreased to 60 seconds, proxy sends options ping - REGISTER is cached and forwarded upstream with Path header and longer expires, i.e. 3600 seconds - UAC reregisters every < 60 seconds but upstream registration is only changed if the timer expires (just less than 3600 seconds) or if AoR changes I good illustration is here: http://www.markholloway.com/blog/?p=1826 My thought is to use htable to store the AoR with an expires just less than 3600, and check it before crafting a new REGISTER with uac module. Is there a better way of handling this scenario? Is there an efficient way populate the $uac_req(…) pseudo variables and call uac_req_send() for multiple, i.e. 4-6 backend servers? I guess all I would need to change is the RURI? Thanks, Spencer

1 0

DBTEXT / AVPOPS and custom database
by Hugh James 12 Nov '13

12 Nov '13

Hi , I'm trying to use the db_text and avpops modules to implement a simple custom database but having problems getting it to work . I'm trying to find the Identity of the recipient for an inbound call based on the called number . I have a database file:- 1) LineIDs - contains the called numbers in the username field and an attribute for that line ID (inbound/ outbound) ,and the recipient identity in the value field I am trying to get the attribute and value pairs that matches the line number in the user name column , id(int,auto) uuid(str,null) username(str) domain(str,null) attribute(str) type(int) value(str) 1::+12125551212::inbound:0:Foster 2::012125551212::inbound:0:Foster 3::12125551212::inbound:0:Foster 4::+4412125551212::inbound:0:Foster 5::+12125551213::inbound:0:Foster 6::012125551213::inbound:0:Foster 7::013440001::inbound:0:Foster 8::call-centre::inbound:0:MBA 9::call-centre::outbound:0:MBA 10::01344665544::inbound:0:chicken 11::01344665544::outbound:0:chicken I try to load the AVPs that match a received number using the following line in the kamailio script file $var(rc)=avp_db_load("01344665544/username","s:recipient/LineIDs"); I have also tried $var(rc)=avp_db_load("01344665544/username","$avp(s:tenants)/LineIDs"); And $var(rc)=avp_db_load("01344665544","$avp(i:123)/LineIDs"); But it fails to find any matching rows ( AVPS) . Am I using the function correctly ? The debug output is shown below , there are some custom debug lines I have added in the db_text C files . DEBUG: avpops [avpops_impl.c:378]: AVPOPS: calls db_load_avp DEBUG: db_text [dbt_base.c:185]: DBTEXT: dbt_query called DEBUG: db_text [dbt_file.c:102]: request for table [LineIDs] DEBUG: db_text [dbt_file.c:109]: db is [/etc/kamailio/Tenants] DEBUG: db_text [dbt_file.c:124]: loading file [/etc/kamailio/Tenants/LineIDs] DEBUG: db_text [dbt_tb.c:187]: mtime is 1383910898 DEBUG: db_text [dbt_file.c:215]: column[0] is INT! DEBUG: db_text [dbt_file.c:210]: column[1] is STR! DEBUG: db_text [dbt_file.c:210]: column[2] is STR! DEBUG: db_text [dbt_file.c:210]: column[3] is STR! DEBUG: db_text [dbt_file.c:210]: column[4] is STR! DEBUG: db_text [dbt_file.c:215]: column[5] is INT! DEBUG: db_text [dbt_file.c:210]: column[6] is STR! DEBUG: db_text [dbt_base.c:240]: new res with 3 cols DEBUG: db_text [dbt_res.c:52]: DBTEXT: dbt_result_new called DEBUG: db_text [dbt_res.c:79]: new res with 3 cols DEBUG: db_text [dbt_res.c:100]: DBTEXT: dbt_result_new returns DEBUG: db_text [dbt_api.c:255]: DBTEXT: dbt_get_result called DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x7f334a85c390 DEBUG: <core> [db_res.c:155]: allocate 24 bytes for result names at 0x7f334a85c3e0 DEBUG: <core> [db_res.c:165]: allocate 12 bytes for result types at 0x7f334a85c1f0 DEBUG: db_text [dbt_api.c:80]: allocate 16 bytes for RES_NAMES[0] at 0x7f334a85c1c0 DEBUG: db_text [dbt_api.c:80]: allocate 16 bytes for RES_NAMES[1] at 0x7f334a85c420 DEBUG: db_text [dbt_api.c:80]: allocate 16 bytes for RES_NAMES[2] at 0x7f334a85c450 DEBUG: db_text [dbt_api.c:281]: DBTEXT: dbt_get_result returns 0 DEBUG: avpops [avpops_impl.c:389]: AVPOPS: db_load_avp returns 0 DEBUG: avpops [avpops_db.c:271]: close avp query DEBUG: <core> [db_res.c:81]: freeing 3 columns DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7f334a85c1c0 DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7f334a85c420 DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7f334a85c450 DEBUG: <core> [db_res.c:94]: freeing result names at 0x7f334a85c3e0 DEBUG: <core> [db_res.c:99]: freeing result types at 0x7f334a85c1f0 DEBUG: <core> [db_res.c:136]: freeing result set at 0x7f334a85c390 DEBUG: avpops [avpops_impl.c:420]: loaded avps = 0 INFO: avp_db_load return result is -1 I was expecting it to return at least one AVP . Any suggestions gratefully received . Hugh James Hugh James | Telephony Developer [icons] * 0845 543 0822 * * Hugh.James(a)semafone.com * www.semafone.com<http://www.semafone.com> * @semafone<http://twitter.com/semafone> * SEMAFONE, 3 The Billings, Walnut Tree Close, Guildford, Surrey GU1 4UL [logo] This e-mail is intended only for the addressee named above. As this e-mail may contain confidential or privileged information if you are not the named addressee or the person responsible for delivering the message to the named addressee, please telephone us immediately. Please note that we cannot guarantee that this message, or any attachment, is virus free or has not been intercepted and amended. The views of the author may not necessarily reflect those of the Company.

2 1

MSILO: remove "Offline message" prefix
by Marc M. 12 Nov '13

12 Nov '13

Hi! I have an up and running msilo config with kamailio 4.0.4. However, all the queued messages are delivered with a prefix "[Offline message - (date)(time)]". Non queued messages are delivered unchanged. Is there a way I can remove this prefix, so also queued messages get delivered unchanged? Please advise! Any help is appreciated! Thanks you, Marc

2 1

rtpproxy-ng ipv4-ipv6 bridge mode problem
by Pavel Miskov 12 Nov '13

12 Nov '13

Hi all, I'm trying to use rtpproxy-ng module (kamailio 4.1 from GIT) and mediaproxy-ng (from https://github.com/sipwise/mediaproxy-ng) in bridge ipv4-ipv6 mode with new NG protocol but keep getting "Failed to extract streams from SDP: No address info found for stream" (log is bellow). Same setup, same SIP clients, same network but with old rtpproxy module and mediaproxy-ng without NG protocol works fine for ipv4-ipv6 bridge mode. of course, only change is in Kamailio configuration where now I load rtpproxy module and change sockets not to use new NG protocol. mediaproxy-ng configuration is untouched. Best regards /Pavel Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Got valid command from 183.24.6.201:57283: answer - { "sdp": "v=0#015#012o=john 0 0 IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012s=-#015#012c=IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012t=0 0#015#012m=audio 5076 RTP/AVP 96 9 97 98 100 102 0 8 103 3 104 101#015#012a=rtpmap:96 opus/48000#015#012a=fmtp:96 usedtx=1#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:97 SILK/24000#015#012a=rtpmap:98 SILK/16000#015#012a=rtpmap:100 speex/32000#015#012a=rtpmap:102 speex/16000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:103 iLBC/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:104 speex/8000#015#012a=rtpmap:101 telephone-event/8000#015#012m=video 5078 RTP/AVP 105 99#015#012a=inactive#015#012a=rtpmap:105 H264/90000#015#012a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1#015#012a=imageattr:105 send * recv [x=[0-1440],y=[0-900]]#015#012a=rtpmap:99 H264/90000#015#012a=fmtp:99 profile-level-id=4DE01f#015#012a=imageattr:99 send * recv [x=[0-1440],y=[0-900]]#015#012", "direction": [ "external", "internal" ], "flags": [ "force", "symmetric" ], "call-id": "60a061ef55655c340a4ffc536383c4ad@0:0:0:0:0:0:0:0", "via-branch": "branch_v4_v6", "received-from": [ "?", "2004:C23:D:CAEF:52e6:BB4:C6F2:1CAD" ], "from-tag": "3a21da16", "to-tag": "5668dfd9", "command": "answer" } Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Failed to extract streams from SDP: No address info found for stream Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Protocol error in packet from 183.24.6.201:57283: Incomplete SDP specification [d3:sdp771:v=0#015#012o=john 0 0 IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012s=-#015#012c=IN IP6 2004:c23:d:caef:3426:666:e36c:12#015#012t=0 0#015#012m=audio 5076 RTP/AVP 96 9 97 98 100 102 0 8 103 3 104 101#015#012a=rtpmap:96 opus/48000#015#012a=fmtp:96 usedtx=1#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:97 SILK/24000#015#012a=rtpmap:98 SILK/16000#015#012a=rtpmap:100 speex/32000#015#012a=rtpmap:102 speex/16000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:103 iLBC/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:104 speex/8000#015#012a=rtpmap:101 telephone-event/8000#015#012m=video 5078 RTP/AVP 105 99#015#012a=inactive#015#012a=rtpmap:105 H264/90000#015#012a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1#015#012a=imageattr:105 send * recv [x=[0-1440],y=[0-900]]#015#012a=rtpmap:99 H264/90000#015#012a=fmtp:99 profile-level-id=4DE01f#015#012a=imageattr:99 send * recv [x=[0-1440],y=[0-900]]#015#0129:directionl8:external8:internale5:flagsl5:force9:symmetrice7:call-id48:60a061ef55655c340a4ffc536383c4ad@0:0:0:0:0:0:0:010:via-branch14:branch_v4_v613:received-froml1:?34:2004:C23:D:CAEF:52e6:BB4:C6F2:1CADe8:from-tag8:3a21da166:to-tag8:5668dfd97:command6:answere] Nov 11 15:58:00 kamailio2 mediaproxy-ng[4790]: Returning to SIP proxy: d6:result5:error12:error-reason28:Incomplete SDP specificatione

2 2

Trunk between Asterisk and Kamailio problem
by Francisco Casasempere 12 Nov '13

12 Nov '13

Hi list, I have this structure IP Phones -> Asterisk Client (LAN) -> (router) -> Kamailio/RTPproxy(Internet) -> Asterisk (local from Kamailio) -> Trunk SIP/Pstn All is working well, I don't have problems with NAT and I managed about 6000 daily calls. For a couple of months I've the next problem.The problem is that in the Asterisk randomly received this message. [Sep 29 04:04:29] NOTICE[3393] chan_sip.c: Peer 'kamONOp' is now Lagged. (2075ms / 2000ms) [Sep 29 04:04:41] NOTICE[3393] chan_sip.c: Peer 'kamailioONO' is now Reachable. (1888ms / 2000ms) In some Asterisk Client I get this message 10 times in a hour, and usually coincide with periods of high load on the kamailo. Also I am getting these messages in the log of Kamailio: Nov 11 08:59:53 c1 /usr/local/sbin/kamailio[8079]: INFO: <core> [forward.c:788]: broken reply to forward - no 2nd via Nov 11 08:59:53 c1 /usr/local/sbin/kamailio[8082]: INFO: <core> [forward.c:788]: broken reply to forward - no 2nd via Are these two issues related? Whay can I do? Kind Regards SiscoCS (Sorry for my english)

2 2

Regd Load Balancer Algorithms
by santosh satapathy 11 Nov '13

11 Nov '13

Hi, I am new to this mailing list and evaluating various options for a good load balancer. Can somebody help let me know if we can choose two algorithms (Ex: weighted and round robin) together for load balancing. -- Thanks Santosh

1 0

SCA Crashes
by Robert Boisvert 11 Nov '13

11 Nov '13

Hi! Looking for some help as to why the SCA module is crashing randomly on 4.0.4. It has crashed several times with a backtrace that shows these stack calls: (gdb) backtrace #0 0x00000000 in ?? () #1 0x4dceca11 in sca_hash_table_slot_kv_delete_unsafe () from /usr/lib/kamailio/modules/sca.so #2 0x4dcecaaa in sca_hash_table_slot_kv_delete () from /usr/lib/kamailio/modules/sca.so #3 0x4dcecb1e in sca_hash_table_index_kv_delete () from /usr/lib/kamailio/modules/sca.so #4 0x4dcecb99 in sca_hash_table_kv_delete () from /usr/lib/kamailio/modules/sca.so #5 0x4dcf30bc in sca_appearance_unregister () from /usr/lib/kamailio/modules/sca.so #6 0x4dd0d9b9 in sca_call_info_update () from /usr/lib/kamailio/modules/sca.so #7 0x126a6a9a in do_action () #8 0x126af88e in run_actions () #9 0x126a4731 in do_action () #10 0x126af88e in run_actions () #11 0x126b007f in run_top_route () #12 0x1269dd7e in receive_msg () #13 0x12734f01 in udp_rcv_loop () #14 0x126d3f56 in main_loop () #15 0x126d76de in main () The message that seemed to trigger the crash in this particular case was the following INVITE: INVITE sip:35705@10.250.16.71;user=phone SIP/2.0 *Record-Route:*<sip:10.250.16.70;lr=on;ftag=D8B14B68-93D21A25;did=513b.a3741> *Via:* SIP/2.0/UDP 10.250.16.70;branch=z9hG4bKc43b.14d180e6.0 *Via:* SIP/2.0/UDP 10.250.88.116;branch=z9hG4bK8d3ec33e632D8AB3 *From:* PIN-1 WKL USA <sip:15862@*obscured*>;tag=D8B14B68-93D21A25 *To:* <sip:0@*obscured*;user=phone> *CSeq:* 1 INVITE *Call-ID:* c69594c-ff17a5d9-d6ed4f6a(a)10.250.88.116 *Contact:* <sip:15862@10.250.88.116> *Allow:* INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER *User-Agent:* PolycomSoundPointIP-SPIP_335-UA/4.0.3.7562 *Accept-Language:* en *Supported:* 100rel,replaces *Allow-Events:* conference,talk,hold *Max-Forwards:* 32 *Content-Type:* application/sdp *Content-Length:* 296 v=0 o=- 1383792680 1383792680 IN IP4 10.250.88.116 s=Polycom IP Phone c=IN IP4 10.250.88.116 t=0 0 a=sendrecv m=audio 2230 RTP/AVP 9 18 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 I have a pcap trace that shows all the calls from 3 hours prior and this is the first time this device appeared. I followed the code down to the sca_hash_table_slot_kv_delete_unsafe ()routine and it appears to have crashed when it tried to use e->free_entry which seems to have been NULL at that point. I'm not sure how it could be NULL since it should have been initialized in sca_hash_table_slot_kv_insert_unsafe (). I'm guessing that either e->free_entry was never initialized or somehow it was overwritten. Since several backtraces end in the same way the bug seems to have be specific to some sequence of events I have not been able to identify. Any help tracking this down would be appreciated. Thanks, Bob

2 2

uninstall kamailio
by Joli Martinez 11 Nov '13

11 Nov '13

Hello, I would like to know how to uninstall Kamailio. I will be upgrading from 3.0 to 3.3 and I would like to uninstall 3.0. I know that there is an upgrade procedure, but 3.0 now does not work and I would like to start from scratch. thanks,

1 0

kamailio dialplan
by Joli Martinez 11 Nov '13

11 Nov '13

I am new to Kamailio and am having an issue with the dialplan setup. I have Kamailio setup as an SBC to handle all user authentication and call routing. I need freeswitch to handle all conferences and voicemails. When I dial 433001 I would like to be transferred to freeswitch for conferences. Right now I have followed the following article and it when I dial 433001 call hangs up and never reaches FS. If I call 43 call does reach FS and I am able to hear FS play the VM prompt. My system is CentoOS 6.4 and FS is installed via yum, but Kamailio is complied. Both FS and Kamailio are on the same box. What commands would you suggest I use to troubleshoot these issues in the future. http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms#dokuwik… Also, since I am new could you give some pointers as far as security and documentation. thanks,

3 4

Re: [SR-Users] tls.so: mod_register failed on fedora 18 x86_64
by Ding Ma 11 Nov '13

11 Nov '13

Hi, all We have applied the openssl 1.0.1e patch described in the email chain between Roberto Fichera and Daniel-Constantin Mierla back in August. But Kamailio 4.0.4 still fails to start up and the log shows memory allocation problem. By the way, we're running kamailio on RHEL 6.5. Wonder if we have missed anything. Please advise. Thanks. Ding

1 0

407 proxy authentication required
by Joli Martinez 11 Nov '13

11 Nov '13

I am setting up Kamailio as an SBC and freeswitch to handle vm and conferences. When I dial 41 Kamailio sends the call to FS and I get the VM prompts, but when I dial 43 the call never hits FS. I am looking at FS in the cli and call never comes in. I get "407 proxy authentication required" from my sip client. I am new to Kamailio and cant seem to figure it out. My system is running CentOS 6.4. FS is installed via yum. Kamailio is complied running version 3.0 thanks,

1 0

implemented specifiers not processed
by Oliver Roth 11 Nov '13

11 Nov '13

Hi all Having some problems with the following part of the script in a timer: We have some kamailios running doing more or less the same job - they are used by a loadbalancer kamailio. Actually I wanted to check the ip or systemname of the current system do perform some actions depending on the used kamailio - based on a timer route[CDRS] { sql_query("ca","call kamailio_cdrs()","rb"); sql_query("ca","call kamailio_rating()","rb"); xlog("timer routine: time is %TF\n"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\n"); xlog("L_INFO", "CDRS- rated Host %Hi"); if %Hi = ipOfGateway1 then do something else if %Hi = ipOfGateway2 then do something different } What I get in log Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR: <script>: timer routine: time is %TF Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR: <script>: time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}> Nov 10 12:54:24 sipgw21 /usr/sbin/kamailio[16231]: INFO: <script>: CDRS- rated Host %Hi The %-variables are not processed ... what am I doing wrong?? Thanks Oli

2 3

kamailio 4.0.4 Record-Route reply setting
by kamailio＠AaronLux.com 11 Nov '13

11 Nov '13

2 1

Carrierroute module
by Oliver Roth 11 Nov '13

11 Nov '13

Hi I get the following error when trying to load the carrierroute module load_module(): ERROR: load_module: could not open module </usr/lib64/kamailio/modules/carrierroute.so>: /usr/lib64/kamailio/modules/carrierroute.so: undefined symbol: cfg_set_error_function I installed Kamailio 4.0.3 by apt-get on Ubuntu 12.04 lts It seems, that with version 3.3 it is working ... root@sipgw22:/home/triotel# kamailio -V version: kamailio 4.0.3 (x86_64/linux) 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, USE_FUTEX, 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, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 17:03:55 Aug 19 2013 with gcc 4.6.3

2 3

BUG? - Dialog module timeout BYE is not ok.
by Uri Shacked 11 Nov '13

11 Nov '13

Hi, Following this issue, and the issue "BYE dialog timeout bad syntax" from Julia. I have 2 kamailio servers. srv1 creates a dlg_timeout BYE and sends it to srv2 that forwards the BYE to the next sip server (some other server...). I can definitely see that the BYE that is being forward is malformed. The TO header info is not complete (the port section for example). When a "normal" bye is received from srv1 (not dlg_timeout) all works fine. One thing that i do in this scenario, on srv2 i use uac_replace() on the TO header. So, it is OK that the final destination is different (i do translations). I attached the BYE from both servers. Any ideas? Thanks, Uri You probably meant the To and From headers, I'm guessing. Yeah, as long as there is no display name component, URIs in headers like that don't have to go in <>s. The way Kamailio sends it is grammatically valid, per RFC 3261. The user agents are at fault for not understanding it correctly. Alex Balashov <abalashov at evaristesys.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >*"BYE headers"? Which headers? *>* *>*Uri Shacked <ushacked at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: *>* *>>*Hi, *>>* *>>*I think there is a bug with the BYE that is sent to the caller and *>>*callee *>>*when dialog timeout happends. *>>* *>> *The BYE headers are sent with no "<" or ">". *>>*So, some sip singaling points decline the BYE. *>>* *>>*BR, *>>*Uri *>>* *>>* *>>*------------------------------------------------------------------------ *>>* *>>*_______________________________________________ *>>*SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing *>*list *>>*sr-users at lists.sip-router.org<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>>*http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users *>* *>*-- *>*Sent from my mobile, and thus lacking in the refinement one might *>*expect from a fully-fledged keyboard. *>* *>*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/ *>* *>*_______________________________________________ *>*SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list * >*sr-users at lists.sip-router.org<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>*http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users * -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. 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/ -------------- next part -------------- An HTML attachment was scrubbed... URL: < http://lists.sip-router.org/pipermail/sr-users/attachments/20130422/0c75272…>

4 11

Wrong "To" header in CANCEL, when "To" has been rewritten by uac_replace_to() in branch route
by Julia 10 Nov '13

10 Nov '13

Hello, I know it was already discussed, that not recommended to modify "To" header and no routing should ever be done on "To". But as long as there are carriers that do it, we need full support for this manipulation for all messages in the call flow. I found in list a few mails with the same problem of wrong header "To" in CANCEL after "To" rewriting. And one solution from Vitaliy Aleksandrov : http://lists.sip-router.org/pipermail/sr-dev/2012-October/016657.html "Kamailio builds "To" header from the received INVITE. Patch to fix it: --- ./kamailio-3.2.3/modules/tm/t_cancel.c 2012-08-22 10:39:21.000000000 +0300 +++ ./kamailio-3.2.3-fixed/modules/tm/t_cancel.c 2012-10-01 18:25:57.000000000 +0300 @@ -284,7 +284,7 @@ if (cfg_get(tm, tm_cfg, reparse_invite)) { /* build the CANCEL from the INVITE which was sent out */ cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, - &t->to + NULL #ifdef CANCEL_REASON_SUPPORT , reason #endif /* CANCEL_REASON_SUPPORT */" We use header "To" manipulation in branch route (after failure route) and this patch solved our problem with "To" in CANCEL. But a patch is not committed. Is it possible to commit this solution? Thank you, Julia

2 4

Modify/Edit INVITE message.
by Aziz 09 Nov '13

09 Nov '13

Hello. I have a question about Kamailio configuration. Can I edit or modify the INVITE message content? For example, remove "To" or "Via" fields in the INVITE content. If yes, how can I do? Thanks.

2 1

#TM: timing problem in serial forking (INVITE vs. CANCEL)
by Klaus Feichtinger 08 Nov '13

08 Nov '13

Hello list, I have troubles with serial forking in kamailio 3.2.4, which is mixed with parallel forking. In the scenario that an initial INVITE message, which is addressed to sip:A, is coming in to the server, it is doing a lookup in the DB and forking (parallel) the request to e.g. 3 SIP user agents. I have set the timer to 20 seconds transaction timeout and after that timeout, the server is handling the original request in the FAILURE_ROUTE[xy]. In this failure route, the request-URI username is overwritten to an alternative one -- e.g. sip:B. Then the server is doing a DB lookup again and forking the request to the number of registered user agents. A specialiy of this scenario is that it can be possible, that user agents have registered for username "A" and username "B" -- in other words: they are members of the parallel forking group in serial forking step 1 and step 2. When the CANCEL and INVITE message would be sent out (to the user agents) in the correct order, then it would be no problem. But in my case the server is sending the "new" INVITE message (2^nd step in the serial forking procedure) to user agents BEFORE the CANCEL request. Therefore, these user agents are rejecting the INVITE message with "500". Signalisation scenario ================== INVITE -> SRV SRV -> INVITE (branch-1) UA1 SRV -> INVITE (branch-2) UA2 SRV -> INVITE (branch-3) UA3 SRV <- 180 (branch-1) UA1 SRV <- 180 (branch-2) UA2 SRV <- 180 (branch-3) UA3 ..... [timeout] SRV -> CANCEL (branch-1) UA1 SRV -> CANCEL (branch-2) UA2 SRV -> INVITE (branch-4) UA4 SRV -> INVITE (branch-5) UA5 SRV -> INVITE (branch-6) UA3 (!!!) SRV -> CANCEL (branch-3) UA3 SRV <- 500 (branch-6) UA3 It was quasi reproduceable that only the last branch of the initial transaction had that timing problem (INVITE <- vs -> CANCEL). My question is: why does the server send the (last) CANCEL message only after the INVITE message for some branch(es)? Could this behaviour be prohibited in any way? Thanks in advance! Klaus

2 2

UUID generation
by Grant Bagdasarian 08 Nov '13

08 Nov '13

Hello, Any plans for building a module which generates a UUID or building it into the core? Daniel once told me to use $sruid, which basically returns a unique value. It works good. I haven't had any duplicates yet, even under heavy load tests.

1 0

kamailio 4.0.4 '408' response
by kamailio＠AaronLux.com 08 Nov '13

08 Nov '13

Below please find a filtered SIP packet capture showing a problem I'm having with callee '408' responses in kamailio 4.0.4. The '408' response occurs after both caller and callee successfully establish 2-way audio. In this setup kamailio is running behind a NAT and both SIP clients connect via WAN and rtpproxy is started with the following command 'rtpproxy -F -l 192.168.1.LAN -A 66.41.221.WAN -s unix:/var/run/rtpproxy.sock -d DBUG:LOG_LOCAL0 -m 10000 -M 20000' Ports 5061 and Ports 10000-20000 are forwarded to the kamailio server and kamailio has TLS enabled. Please let me know if you know if you need more info. Thanks, Aaron # Filtered SIP capture showing only 408 response: http://pastebin.com/pBcZNGA4 # rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Extension 20090810: Support for automatic bridging # kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 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, USE_FUTEX, 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, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7

2 1

1 via header
by phillman25 08 Nov '13

08 Nov '13

Dear List I am trying to interconnect with a Cisco Call manager express, i am sending the INVITE message with the below format: INVITE sip:22775019@81.21.39.153 SIP/2.0 Record-Route: <sip:81.21.38.33;lr=on;ftag=685329680;did=671.51a2> Via: SIP/2.0/UDP 81.21.38.33;branch=z9hG4bKda1d.291eab1622fb619ec7df3ca647581a19.0 Via: SIP/2.0/UDP 81.21.35.22:5060;branch=z9hG4bKterm-4a32-99665015-6855A22775019-37284 From: 99665015 <sip:99665015@81.21.35.22;user=phone>;tag=685329680 To: 6855A22775019 <sip:6855A22775019@81.21.38.33;user=phone> Call-ID: 6fea2437-2d338fcf-4efc6b90-2d08(a)81.21.35.22 CSeq: 1 INVITE Supported: timer Session-Expires: 1800 Min-SE: 1800 Contact: <sip:99665015@81.21.35.22:5060> Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Max-Forwards: 69 Content-Type: application/sdp Content-Length: 528 P-hint: outbound v=0 o=- 3099173 0 IN IP4 81.21.35.44 s=Cisco SDP 0 c=IN IP4 81.21.35.44 t=0 0 m=audio 52168 RTP/AVP 0 8 18 99 102 103 104 4 105 3 106 107 108 125 101 100 a=rtpmap:99 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=nortpproxy:yes The Cisco Manager Express replies by sending the TRYING message in the below format with 1 via header: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.21.38.33;branch=z9hG4bKda1d.291eab1622fb619ec7df3ca647581a19.0,SIP/2.0/UDP 81.21.35.22:5060 ;branch=z9hG4bKterm-4a32-99665015-6855A22775019-37284 From: 99665015 <sip:99665015@81.21.35.22;user=phone>;tag=685329680 To: 6855A22775019 <sip:6855A22775019@81.21.38.33;user=phone> Date: Tue, 05 Nov 2013 18:24:17 GMT Call-ID: 6fea2437-2d338fcf-4efc6b90-2d08(a)81.21.35.22 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Seems like Kamailio is not accepting this, is the Trying message from the client correct? Is it correct that he is sending 1 via header with 2 IP's as opposed to 2 via headers with 1 IP in each via header? Otherwise is there a solution for this? Thanks in advance Phillip

2 1

Where do I place the logic for modifying auth and call requests?
by erik＠valerius.se 08 Nov '13

08 Nov '13

Hello everyone. Some of my confusion probably comes from me coming from the web development world and not software dev, but I would greatly appreciate any response to the following question: I use the MySQL auth for my subscribers which is working great. However, let's say that I add the column "valid_until" in the subscriber table and want to check if today's date is less than the caller's valid_until value. I can set up the sqlops module without a problem and get the actual value, but I have no idea where I place the logic to compare it to a date and then pass on the OK message to Kamailio to continue the request. Is this done with embedded scripts? Or right there in the config file? Where can I find a list of what parts of the call request are exposed for customization from Kamailio? I'd greatly prefer to do things in Python or JavaScript, if that matters. Many thanks in advance! Erik Eklund

2 2

RTPproxy support in Wireshark
by Peter Lemenkov 08 Nov '13

08 Nov '13

Hello All! First of all I'm sorry for the crossposting. For those who unaware of what's going inside Wireshark project - I'm working on a "classic" RTPproxy protocol dissector. Most of the patches already accepted, and Wireshark's RTPproxy parser is already usable. Few more patches should land in trunk quite soon. As for rtpproxy-ng - I plan to add support for its protocol as well (right after I finish with this one). If you're using Fedora Linux, then Wireshark from the repository already compiled with RTPproxy protocol dissection support. -- With best regards, Peter Lemenkov.

1 0

Mitigation of unavailable rtpproxy
by Alex Balashov 07 Nov '13

07 Nov '13

Hello, (Sorry for cross-posting to -users and -dev; not really sure where this post belongs most.) A few days ago, I ran into an issue with a Kamailio server being somewhat unresponsive, during moderate call volume, on account of a down rtpproxy--the only rtpproxy in the set. This is rtpproxy classic, not ngcp-mediaproxy-ng. Rtpproxy was not actually engaged on any of the initial INVITEs going through the server; the server is configured to invoke it conditionally based on a setting, and the setting was not set for any endpoints. rtpproxy_manage() was never called. However, I call unforce_rtp_proxy() unconditionally in my config when handling CANCELs, reasoning that it can't do any harm if rtpproxy_manage() was not called before[1]. Nevertheless, it seemed to be the case that this situation was clogging up SIP worker threads, because some SIP messages were definitely dropped. Periodic log messages about inability to reach the rtpproxy were echoed as well. This problem cleared up almost immediately when the rtpproxy instance was restored into service. This raised some questions in my mind about the relationship between rtpproxy management and SIP worker thread utilisation. I assume it was my indiscriminate unforce_rtp_proxy() calls that were actually clogging up the worker threads, right? If so, why? I figured that in the unforce_rtp_proxy() case, the rtpproxy module simply sends fire-and-forget UDP messages down the UDP control socket without any sort of blocking for acknowledgement, since in this case the call must be released on the rtpproxy side without doing any rewriting of SDP on the Kamailio side (unlike in the case where rtpproxy is engaged). Thus, there should be no need to wait for ports to substitute into the message. Or is the same response-wait mechanism used regardless, even in the unforce_rtp_proxy() case, for programmatic reasons? More broadly, is there any way that this scenario can be prevented? In other words, is there a way to work around an outage of all rtpproxies in the set without tying up workers, or at least tying them up less severely? Thanks! -- Alex [1] Is this a reasonable assumption? The reason I do this is that I don't see a way to find out if rtpproxy was engaged from the body of a CANCEL message. I do check for a ;proxy_media RR parameter when handling BYEs, but since a CANCEL is not an in-dialog request, I'm not sure what to do except to call unforce_rtp_proxy()/rtpproxy_manage() indiscriminately, without resorting to storing state in htable or other complications I don't want. -- 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/

3 5

Kamctl ping works as my unix user, not in root nor kamailio user
by Benjamin Henrion 07 Nov '13

07 Nov '13

Hi, I am trying to find out why kamctl ping sip:1.2.3.4:5060 works with my unix user (got a 200 OK), while it does not with kamailio nor root user (I got a 408 timeout). Any idea? The permissions of the socket in /tmp are like this: ======================================= srw------- 1 root root 0 Oct 31 01:01 kamailio_ctl prw-rw---- 1 kamailio kamailio 0 Nov 7 13:55 kamailio_fifo srw-rw-rw- 1 kamailio kamailio 0 Oct 31 01:01 kamailio_socket ======================================= Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

2 1

SIP registration
by Mudasser 07 Nov '13

07 Nov '13

Hello I am devloping the chat/call application like viber,whatsapp etc in Android using sip. I want to know how i can register the sip account with ur sevrer. what's code which i need to write in my application.suppose if i want to chat or call between other person how i can get register without telling user that i am going for sip registration means there i cam not going to display sip registration form etc. where i can save user message and how i can maintain user info like dispaly name , display pic etc.is there any serverside databse involvement to save messages,pic etc or sip sever maintain automatically. Kindly describe in details as i am new in Android Thank you Best Regards Mudasser Tufail Senior Software Engineer

1 0

Re: [SR-Users] #cfgutils: sleep(x) does not work in failure_route
by Klaus Feichtinger 07 Nov '13

07 Nov '13

Hello Daniel, I´ve patched the code and found that the return value of the 'sleep' command is - as expected - differing in the scenarios. For scenario 1 (no provisional response), the return value is '0' and for scenario 2, the return value is '2' (see log output): scenario 1: Nov 7 12:40:51 sipsrvnode1 kamailio33[10974]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 7 12:40:51 sipsrvnode1 kamailio33[10974]: DEBUG: cfgutils [cfgutils.c:650]: sleep 2 seconds Nov 7 12:40:53 sipsrvnode1 kamailio33[10974]: DEBUG: cfgutils [cfgutils.c:652]: sleep-rc 0 Nov 7 12:40:53 sipsrvnode1 kamailio33[10974]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s scenario 2: Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: DEBUG: cfgutils [cfgutils.c:650]: sleep 2 seconds Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: DEBUG: cfgutils [cfgutils.c:652]: sleep-rc 2 Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s This means that the sleep command is interrupted (in scenario 2) by a signal, as the return value is representing the "number of seconds left to sleep, if the call was interrupted by a signal handler.". Which signal is interrupting "sleep" in this scenario (only) :-( ? Klaus > Daniel-Constantin Mierla <miconda(a)gmail.com> hat am 7. November 2013 um 10:36 > geschrieben: > > Hello, > > can you try to see if sleep_us() works? > > It is rather strange because sleep() from cfgutils just uses sleep() from > system library, nothing special internally... sleep is interrupted by signals > as well. Can you patch the function in kamailio to log the return code from > sleep()? > > Cheers, > Daniel > > On 11/7/13 8:40 AM, klaus.lists#inode.at wrote: > > > > Hello, > > > > I have an update: > > > > today I´ve tested with kamailio version 4.0.4, but the behaviour is > > still the same: it is working fine only when no response has been received; > > as soon as a provisional response is received, the failure route is ignoring > > the "sleep" function. > > > > Klaus > > > > > > > Hello list, > > > > > > I have troubles using the function "usleep(x)" or "sleep(x)" (the > > > behaviour is the same) of the "cfgutils" module > > > (<http://kamailio.org/docs/modules/3.2.x/modules_k/cfgutils.html#idp76968> > > > ) in the failure_route. According documentation, this function could be > > > used in any route, including the failure_route, too. However, when I call > > > this function, it does not show any reaction, if a transaction was already > > > answered with a provisional response. It is still working fine, when a > > > transaction is timing out, as no target socket is reachable. This is > > > confusing me! > > > > > > Please find below some xlog messages from my configuration file in > > > two scenarios (function "t_set_fr(20000, 10000);" is identic in both > > > scenarios): > > > (1) scenario 1 is including a primary target, looked up in the > > > registrar DB, which is unreachable; after transaction timeout I have > > > inserted the function "sleep()" in the failure_route for forcing a delay > > > => here it is working fine (see timestamp) > > > > > > (2) scenario 2 is including a primary target which is responding, > > > but not establishing the session; after transaction timeout the procedure > > > is still the same as in scenario (1) > > > => here is no delay visible in the log and practically detectable > > > > > > From my point of view, this is a malfuntion! Does anybody see it > > > different? I´ve tested these scenarios with kamailio-3.2.x and > > > kamailio-3.3.x - no difference. > > > > > > Thanks for any hints! > > > > > > Klaus > > > > > > ################# SCENARIO (1) ORIG TARGET IS UNREACHABLE > > > ########################## > > > Nov 6 15:58:25 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: > > > -<|XLOG|>-: <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > > > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > > > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > > > > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 > > > From:<sip:1101015555@10.16.48.71> To:<sip:115300@10.16.48.44> > > > > > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a > > > BRANCH_TIMEOUT of RU:<sip:117002@10.16.48.226:5678> > > > > > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s > > > > > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s > > > > > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: > > > -<|XLOG|>-: <RELAYFB> is reached for RU:<sip:117003@10.16.48.226:7001> , > > > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > > > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > > > > > ################# SCENARIO (2) ORIG TARGET IS REACHABLE > > > ########################## > > > > > > Nov 6 16:02:14 sipsrvnode1 /usr/sbin/kamailio[29034]: INFO: > > > -<|XLOG|>-: <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > > > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > > > INVITE, Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 > > > From:<sip:5555@10.16.48.71> To:<sip:4000@10.16.48.44> > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a > > > BRANCH_TIMEOUT of RU:<sip:117002@10.16.48.44> > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <RELAYFB> is reached for RU:<sip:117003@10.16.48.44> , > > > From:<sip:5555@10.16.48.71> , To:<sip:4000@10.16.48.44> , Method: INVITE, > > > Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> > > > > > > > > > > > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org <mailto:sr-users@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> > Kamailio Advanced Trainings - Berlin, Nov 25-28 > - more details about Kamailio trainings at <http://www.asipto.com> - >

1 0

IM Broadcasting
by sipatse＠in.gr 07 Nov '13

07 Nov '13

1 0

Function to test DB connection from route[]?
by Allen Zhang 07 Nov '13

07 Nov '13

Hi all, Is there a function I can use to test database connection from the routing script in the cfg file? My dispatcher uses OPTIONS probing. When the destination receives the OPTIONS request, I'd like to detect the health status of the database connection before sending a 200 back. Note the OPTIONS request doesn't have a user part in the RURI. Hence, is_subscriber() and alias_db_lookup() don't work. Any suggestion? Regards, Allen Zhang

2 1

Re: [SR-Users] #cfgutils: sleep(x) does not work in failure_route
by klaus.lists#inode.at 07 Nov '13

07 Nov '13

Hello, I have an update: today I´ve tested with kamailio version 4.0.4, but the behaviour is still the same: it is working fine only when no response has been received; as soon as a provisional response is received, the failure route is ignoring the "sleep" function. Klaus > Hello list, > > I have troubles using the function "usleep(x)" or "sleep(x)" (the behaviour > is the same) of the "cfgutils" module > (<http://kamailio.org/docs/modules/3.2.x/modules_k/cfgutils.html#idp76968> ) > in the failure_route. According documentation, this function could be used in > any route, including the failure_route, too. However, when I call this > function, it does not show any reaction, if a transaction was already answered > with a provisional response. It is still working fine, when a transaction is > timing out, as no target socket is reachable. This is confusing me! > > Please find below some xlog messages from my configuration file in two > scenarios (function "t_set_fr(20000, 10000);" is identic in both scenarios): > (1) scenario 1 is including a primary target, looked up in the registrar DB, > which is unreachable; after transaction timeout I have inserted the function > "sleep()" in the failure_route for forcing a delay > => here it is working fine (see timestamp) > > (2) scenario 2 is including a primary target which is responding, but not > establishing the session; after transaction timeout the procedure is still the > same as in scenario (1) > => here is no delay visible in the log and practically detectable > > From my point of view, this is a malfuntion! Does anybody see it different? > I´ve tested these scenarios with kamailio-3.2.x and kamailio-3.3.x - no > difference. > > Thanks for any hints! > > Klaus > > ################# SCENARIO (1) ORIG TARGET IS UNREACHABLE > ########################## > Nov 6 15:58:25 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: > <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > <FAILRELAY> is reached with Code: 408 From:<sip:1101015555@10.16.48.71> > To:<sip:115300@10.16.48.44> > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of > RU:<sip:117002@10.16.48.226:5678> > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e e p 2000ms / 2s > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e p t 2000ms / 2s > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: > <RELAYFB> is reached for RU:<sip:117003@10.16.48.226:7001> , > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > ################# SCENARIO (2) ORIG TARGET IS REACHABLE > ########################## > > Nov 6 16:02:14 sipsrvnode1 /usr/sbin/kamailio[29034]: INFO: -<|XLOG|>-: > <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > INVITE, Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <FAILRELAY> is reached with Code: 408 From:<sip:5555@10.16.48.71> > To:<sip:4000@10.16.48.44> > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of > RU:<sip:117002@10.16.48.44> > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e e p 2000ms / 2s > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e p t 2000ms / 2s > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <RELAYFB> is reached for RU:<sip:117003@10.16.48.44> , > From:<sip:5555@10.16.48.71> , To:<sip:4000@10.16.48.44> , Method: INVITE, > Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> >

2 1

Email alert
by phillman25 07 Nov '13

07 Nov '13

Hi I was wondering which module i could use to configure an email alert upon an event in kamailio.cfg file i.e. if ...... then send to email address xxxx any ideas? thanks! Phillip

3 2

IM Broadcasting
by sipatse＠in.gr 07 Nov '13

07 Nov '13

1 0

408 callee hangup problem
by kamailio＠AaronLux.com 06 Nov '13

06 Nov '13

2 3

Google search
by Carlos Ruiz Díaz 06 Nov '13

06 Nov '13

Hi guys, this happens to me all the time: when I search for documentation about a specific Kamailio module on Google, I always land on outdated entries usually from Kamailio 1.6.x. Although this seems trivial and I could simply update the URL to the current version of Kamailio, this may be not so obvious for new users and they may end up reading expired documentation and/or implementing non-optimal solutions to their problems. is there a way to only allow indexation of the last documentation? Something like removing the version or adding an HTTP 301 to old entries? Regards, -- Carlos http://caruizdiaz.com +595981146623

3 2

#cfgutils: sleep(x) does not work in failure_route
by Klaus Feichtinger 06 Nov '13

06 Nov '13

Hello list, I have troubles using the function "usleep(x)" or "sleep(x)" (the behaviour is the same) of the "cfgutils" module (http://kamailio.org/docs/modules/3.2.x/modules_k/cfgutils.html#idp76968) in the failure_route. According documentation, this function could be used in any route, including the failure_route, too. However, when I call this function, it does not show any reaction, if a transaction was already answered with a provisional response. It is still working fine, when a transaction is timing out, as no target socket is reachable. This is confusing me! Please find below some xlog messages from my configuration file in two scenarios (function "|t_set_fr(20000, 10000);" is identic in both scenarios|): (1) scenario 1 is including a primary target, looked up in the registrar DB, which is unreachable; after transaction timeout I have inserted the function "sleep()" in the failure_route for forcing a delay => here it is working fine (see timestamp) (2) scenario 2 is including a primary target which is responding, but not establishing the session; after transaction timeout the procedure is still the same as in scenario (1) => here is no delay visible in the log and practically detectable From my point of view, this is a malfuntion! Does anybody see it different? I´ve tested these scenarios with kamailio-3.2.x and kamailio-3.3.x - no difference. Thanks for any hints! Klaus ################# SCENARIO (1) ORIG TARGET IS UNREACHABLE ########################## Nov 6 15:58:25 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: <RELAY> is reached for RU: sip:117002@10.16.48.226:5678, From: sip:1101015555@10.16.48.71, To: sip:115300@10.16.48.44, Method: INVITE, Call-ID: 1107651805(a)10.16.48.71 Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 From: sip:1101015555@10.16.48.71 To: sip:115300@10.16.48.44 Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of RU: sip:117002@10.16.48.226:5678 Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: <RELAYFB> is reached for RU: sip:117003@10.16.48.226:7001, From: sip:1101015555@10.16.48.71, To: sip:115300@10.16.48.44, Method: INVITE, Call-ID: 1107651805(a)10.16.48.71 ################# SCENARIO (2) ORIG TARGET IS REACHABLE ########################## Nov 6 16:02:14 sipsrvnode1 /usr/sbin/kamailio[29034]: INFO: -<|XLOG|>-: <RELAY> is reached for RU: sip:117002@10.16.48.226:5678, From: sip:1101015555@10.16.48.71, To: sip:115300@10.16.48.44, Method: INVITE, Call-ID: 553330101(a)10.16.48.71 Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 From: sip:5555@10.16.48.71 To: sip:4000@10.16.48.44 Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of RU: sip:117002@10.16.48.44 Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <RELAYFB> is reached for RU: sip:117003@10.16.48.44, From: sip:5555@10.16.48.71, To: sip:4000@10.16.48.44, Method: INVITE, Call-ID: 553330101(a)10.16.48.71

1 0

dispatcher: ds_options_callback, Setting the state failed
by Seudin Kasumovic 06 Nov '13

06 Nov '13

Hi, Test configuration uses OPTION for ping method. In log below host 198.51.100.68 probed first but reply come latter then 192.0.2.4. Seams that only last probed uri tried for setting state other hosts. Also, DEBUG log is broken. (kamailio version: kamailio 4.1.0-pre1 (i386/linux)). Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: dispatcher [dispatch.c:2420]: ds_check_timer(): probing set #28, URI sip: 198.51.100.68:5060 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:172.16.23.130> Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:182]: dlg2hash(): DEBUG: dlg2hash: 33871 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:345]: t_uac_prepare(): executing event_route[tm:local-request] Nov 5 12:48:01 localhost kamailio[23729]: INFO: <script>: 1280:tm:local-request OPTIONS sip:198.51.100.68:5060 --ID= 77db7ea0-23729(a)172.16.23.5 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:410]: t_uac_prepare(): apply new updates to sip msg Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: dispatcher [dispatch.c:2420]: ds_check_timer(): probing set #0, URI sip:192.0.2.4:5090 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:172.16.23.130> Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:182]: dlg2hash(): DEBUG: dlg2hash: 33872 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:345]: t_uac_prepare(): executing event_route[tm:local-request] Nov 5 12:48:01 localhost kamailio[23729]: INFO: <script>: 1280:tm:local-request OPTIONS sip:192.0.2.4:5090 --ID= 77db7ea1-23729(a)172.16.23.5 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=54 global id=53 T start=(nil) Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 33871 label 0 branch 0 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb492a8a8)! Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=54 global id=54 T end=0xb492a8a8 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:2205]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=2 is_invite=0) Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:1309]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=200 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:2085]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:2122]: local_reply(): DEBUG: local transaction completed Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb492a8a8, callback type 1024, id 0 entered Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: dispatcher [dispatch.c:2359]: ds_options_callback(): OPTIONS-Request was finished with code 200 (to 192.0.2.4:5090> Nov 5 12:48:01 localhost kamailio[23725]: ERROR: dispatcher [dispatch.c:2372]: ds_options_callback(): Setting the state failed ( 192.0.2.4:5090> Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:1667]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=50 global id=49 T start=(nil) Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 33872 label 0 branch 0 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb492c918)! Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=50 global id=50 T end=0xb492c918 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:2205]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=2 is_invite=0) Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:1309]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=200 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:2085]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:2122]: local_reply(): DEBUG: local transaction completed Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb492c918, callback type 1024, id 0 entered Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: dispatcher [dispatch.c:2359]: ds_options_callback(): OPTIONS-Request was finished with code 200 (to sip:192.0.2.4:5090, group 0) ds_list shows: SET_NO:: 2 SET:: 28 URI:: sip:198.51.100.68:5060 flags=TX priority=1 attrs= SET:: 0 URI:: sip:192.0.2.4:5090 flags=AP priority=1 attrs= -- Seudin Kasumovic

2 2

use append_hf() and remove_hf() on the same message
by Camille Oudot 06 Nov '13

06 Nov '13

Hi, I'm using remove_hf() and append_hf() to manipulate the sip headers on my application. Apparently, calling insert_hf("foo") after remove_hf("bar") if bar is the last header (not including Content-Length) is a bad idea, foo doesn't get inserted (I understand that the append_hf() command tries to insert foo after bar, and since bar just disappeared, well, it fails) What is the safest way to both remove and append headers on the same message? Regards -- Camille

2 4

dialog stats reported by app_perl
by mayamatakeshi 06 Nov '13

06 Nov '13

Hello, I am using kamailio 4.0. I am experimenting with module app_perl. Actually, I am not using it, I am just loading it because it solves this problem when loading snmpstats: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader However, before using app_perl i was seeing this: # kamctl fifo get_statistics dialog active_dialogs dialog:active_dialogs = 1 But with app_perl loaded, I see this: # kamctl fifo get_statistics dialog active_dialogs app_perl:active_dialogs = 1 So, is app_perl intercepting stats requests that should be handled by module dialog? Is this expected? And, is there any performance penalty on having this handled by perl? Regards, Takeshi

2 2

TLS - No ServerHello
by Coy Cardwell 05 Nov '13

05 Nov '13

Hi all. We are using Kamailio 4.0.4 in an Edge Proxy to Central Registrar configuration. The system is in a functional state. Oddly, when we enable TLS, the Kamailio server never responds with a Server Hello to the Client Hello for TLS, so a connection is never established. Has anyone seen anything like this? There are no errors anywhere, the server just never responds to the initial TLS Client Hello. We know the packet arrives and is acknowledged at the TCP level. Just a bit stumped.

2 2

Question about call transfer and pickup
by Гергарт Олег (Мобилон) 05 Nov '13

05 Nov '13

Hello! Tell me please, is there in kamailio opportunity to call transfer and pickup? If yes, where can I about this to read? Thanks! Regards, Oleg Gergart

2 1

Segmentaion fault in Kamailio 4.0.3
by Morten Isaksen 05 Nov '13

05 Nov '13

Hi, Our Kamailio has stopped with a segmentation fault 4 times the last week. There has been no changes to the configuration file the last 15 days, so I suspect a SIP phone is sending a SIP packet that Kamailio does not like. We have a core dump file but I cannot read anything usefull from the backtrace. Can you see what is wrong from the backtrace? Regards Morten The output from gdb (bt full): Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 256 -M 8 -u kamai'. Program terminated with signal 11, Segmentation fault. #0 0x00007f0e10de17b2 in cancel_branch (t=0x7f0dfbf38e10, branch=0, reason=<value optimized out>, flags=4) at t_cancel.c:284 284 if (cfg_get(tm, tm_cfg, reparse_invite) || Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 hiredis-0.10.1-3.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.10.3-10.el6.x86_64 libcom_err-1.41.12-14.el6.x86_64 libselinux-2.0.94-5.3.el6.x86_64 libxml2-2.7.6-12.el6_4.1.x86_64 mysql-libs-5.1.67-1.el6_3.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-27.el6_4.2.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt full #0 0x00007f0e10de17b2 in cancel_branch (t=0x7f0dfbf38e10, branch=0, reason=<value optimized out>, flags=4) at t_cancel.c:284 cancel = <value optimized out> len = <value optimized out> crb = 0x7f0dfbf39008 irb = 0x7f0dfbf38f80 ret = 1 tmp_cd = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 0}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 0}}}} pcbuf = <value optimized out> __FUNCTION__ = "cancel_branch" #1 0x00007f0e10e298ab in reply_received (p_msg=0x7f0e124ce760) at t_reply.c:2194 msg_status = <value optimized out> last_uac_status = 408 ack = 0x7f0dfbf38e10 "Ð \267\373\r\177" ack_len = <value optimized out> branch = 0 reply_status = <value optimized out> onreply_route = <value optimized out> cancel_data = {cancel_bitmap = 0, reason = {cause = 408, u = {text = {s = 0x0, len = 307468800}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 307468800}}}} uac = <value optimized out> t = 0x7f0dfbf38e10 lack_dst = {send_sock = 0x0, to = {s = {sa_family = 6704, sa_data = "\\#\377\177\000\000\000\000\000\000\000\000\000"}, sin = {sin_family = 6704, sin_port = 9052, sin_addr = {s_addr = 32767}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 6704, sin6_port = 9052, sin6_flowinfo = 32767, sin6_addr = {__in6_u = { __u6_addr8 = "\000\000\000\000\000\000\000\000p{1\022\016\177\000", __u6_addr16 = {0, 0, 0, 0, 31600, 4657, 32526, 0}, __u6_addr32 = {0, 0, 305232752, 32526}}}, sin6_scope_id = 307029856}}, id = 32526, proto = 28 '\034', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} backup_user_from = <value optimized out> backup_user_to = <value optimized out> backup_domain_from = <value optimized out> backup_domain_to = <value optimized out> backup_uri_from = <value optimized out> backup_uri_to = <value optimized out> backup_xavps = <value optimized out> replies_locked = 0 branch_ret = <value optimized out> prev_branch = <value optimized out> blst_503_timeout = <value optimized out> hf = <value optimized out> onsend_params = {req = 0x8d8a39, rpl = 0x541db4, param = 0x7f0e1253e528, code = 307029856, flags = 32526, branch = 0, t_rbuf = 0x7fff235c1a30, dst = 0x7f0e12317b70, send_buf = { s = 0x375311000000000 <Address 0x375311000000000 out of bounds>, len = 0}} ctx = {rec_lev = 307491008, run_flags = 32526, last_retcode = 5674412, jmp_env = {{__jmpbuf = {140733786626256, 63331951475841423, 139698413054576, 139698413299552, 9276465, 139698039855608, -63254168797292145, 63332490682325391}, __mask_was_saved = 0, __saved_mask = {__val = {139698413734928, 0, 139698411522821, 1, 140733786626608, 6185835, 5972697, 8586176, 9275699, 69026945952, 3, 9276465, 9275673, 139698413738496, 9275961, 139698413760704}}}}} __FUNCTION__ = "reply_received" #2 0x0000000000456444 in do_forward_reply (msg=0x7f0e124ce760, mode=<value optimized out>) at forward.c:799 new_buf = 0x0 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} new_len = <value optimized out> r = <value optimized out> s = <value optimized out> len = <value optimized out> __FUNCTION__ = "do_forward_reply" #3 0x000000000049e15e in receive_msg (buf=<value optimized out>, len=313, rcv_info=0x7fff235c1cd0) at receive.c:270 msg = 0x7f0e124ce760 ctx = {rec_lev = 11, run_flags = 0, last_retcode = 206110737, jmp_env = {{__jmpbuf = {139698036884436, 11, 219309716216, 139698419720192, 140733786627520, 4294967295, 140733786627647, 1}, __mask_was_saved = 8576456, __saved_mask = {__val = {0, 28, 16, 0, 219305533392, 1, 0, 139698411461552, 219309716216, 139698036884436, 139698413732672, 139698419717800, 139698413732680, 140733786627416, 219305559701, 140733786627288}}}}} ret = <value optimized out> inb = { s = 0x8d8900 "SIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 178.21.249.20;branch=z9hG4bK8149.c6575a95.0\r\nTo: sip:201@78799865.pbx.one-connect.dk;tag=07c44e68\r\nFrom: sip:201@78799865.pbx.one-connect.dk;tag=a6a1c5f60faecf035a"..., len = 313} __FUNCTION__ = "receive_msg" #4 0x0000000000530e46 in udp_rcv_loop () at udp_server.c:557 ---Type <return> to continue, or q <return> to quit--- len = 313 buf = "SIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 178.21.249.20;branch=z9hG4bK8149.c6575a95.0\r\nTo: sip:201@78799865.pbx.one-connect.dk;tag=07c44e68\r\nFrom: sip:201@78799865.pbx.one-connect.dk;tag=a6a1c5f60faecf035a"... from = 0x7f0e12538340 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {2993962576, 0}, addr32 = {2993962576, 0, 0, 0}, addr16 = {15952, 45684, 0, 0, 0, 0, 0, 0}, addr = "P>t\262", '\000' <repeats 11 times>}}, dst_ip = { af = 2, len = 4, u = {addrl = {351868338, 0}, addr32 = {351868338, 0, 0, 0}, addr16 = {5554, 5369, 0, 0, 0, 0, 0, 0}, addr = "\262\025\371\024", '\000' <repeats 11 times>}}, src_port = 35754, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\213\252P>t\262\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 43659, sin_addr = {s_addr = 2993962576}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 43659, sin6_flowinfo = 2993962576, sin6_addr = {__in6_u = { __u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x7f0e124cfbd0, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #5 0x000000000046716a in main_loop () at main.c:1638 i = <value optimized out> pid = <value optimized out> si = <value optimized out> si_desc = "udp receiver child=2 sock=178.21.249.20:5060 \000\000\000\000\200\303P\022\016\177\000\000\000\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000@\350\216\000\000\000\000\000\001\000\000\000\000\000\000\000\200\350\216\000\000\000\000\000\000\000\200\020", '\000' <repeats 12 times>, "\005\000\000\000\000\000\000" nrprocs = <value optimized out> __FUNCTION__ = "main_loop" #6 0x000000000046a002 in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566 cfg_stream = <value optimized out> c = <value optimized out> r = <value optimized out> tmp = 0x7fff235c377f "" tmp_len = 0 options = 0x5c08c8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1722854551 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = <value optimized out> n_lst = <value optimized out> p = <value optimized out> __FUNCTION__ = "main" (gdb) -- Morten Isaksen

2 9

Undefined Symbol Error in TLS.so
by Mark Zeman 05 Nov '13

05 Nov '13

Hi, I'm trying to set up a Kamailio server for a project at university, and have some trouble with it. Since Kamailio isn't starting, I use the 'kamailio -c -f kamailio.cfg' command to figure out what's wrong. This tells me that it cannot load the tls.so file, because of 'undefined symbol: log_color. I've tried to recompile the TLS module, but that hasn't helped. Does anyone know how to fix this? Cheers, Mark

2 4

Callee timeout
by kamailio＠AaronLux.com 05 Nov '13

05 Nov '13

The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out. Network Topography: (ip addresses were changed to make them easier to read.) 66.87.78.CALLER is Caller running csipsimple 66.87.76.CALLEE is Callee running csipsimple 192.168.1.LAN is LAN ip running rtpproxy and kamailio 66.41.221.WAN is WAN ip with ports 5004-5082 forwarded to 192.168.1.LAN And firewall turned off. ###################### kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 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, USE_FUTEX, 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, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7 KAMAILIO DEBUG LOG: 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=472 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=577 a=25 n=mf_process_maxfwd_header 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=587 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=26 n=sanity_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=475 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=765 a=55 n=force_rport 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=766 a=25 n=nat_uac_test 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=771 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=767 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=add_contact_alias 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=772 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=478 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=631 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=591 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=491 a=24 n=t_check_trans 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=752 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=739 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=747 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=742 a=27 n=auth_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=743 a=26 n=auth_challenge 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=744 a=2 n=exit 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=472 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=577 a=25 n=mf_process_maxfwd_header 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=587 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=26 n=sanity_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=475 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=765 a=55 n=force_rport 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=766 a=25 n=nat_uac_test 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=771 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=767 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=add_contact_alias 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=772 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=478 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=631 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=591 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=491 a=24 n=t_check_trans 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=752 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=739 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=747 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=742 a=27 n=auth_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=749 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=747 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=748 a=24 n=consume_credentials 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=759 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=759 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=25 n=remove_hf 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=503 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=499 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=500 a=24 n=record_route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=509 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=503 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=505 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=509 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=826 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=514 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=719 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=694 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=695 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=517 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=648 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=635 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=527 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=527 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=860 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=530 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=666 a=63 n=assign 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=683 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=667 a=25 n=lookup 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=688 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=683 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=685 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=688 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=541 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=538 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=539 a=25 n=t_is_set 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=539 a=25 n=t_on_branch 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=544 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=541 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=543 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=542 a=25 n=t_is_set 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=542 a=25 n=t_on_reply 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=548 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=544 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=546 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=545 a=25 n=t_is_set 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=545 a=25 n=t_on_failure 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=551 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=548 a=24 n=t_relay 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=907 a=25 n=xdbg 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=908 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=787 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=782 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 52(3514) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=794 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=798 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=795 a=24 n=t_is_branch_route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=796 a=25 n=add_rr_param 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=907 a=25 n=xdbg 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=908 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=787 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=782 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 52(3514) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=794 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=798 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=795 a=24 n=t_is_branch_route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=796 a=25 n=add_rr_param 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=551 a=2 n=exit 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1383447347;method=INVITE;from_tag=ghp-tpel0iRnq1kdZVv2Ovu0HuNDES6J;to_tag=nZnsaAYjbOaSx2u-knYbcLDvQh2ymtiI;call_id=mh5qsRvw4rQGbPMmJS2oS06Dsof89co6;code=200;reason=OK;src_user=5465;src_domain=66.41.221.WAN;src_ip=66.87.78.CALLER;dst_ouser=5466;dst_user=5466;dst_domain=66.87.76.CALLEE 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 54(3519) ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 192.168.1.165:43177 failed 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return ################## rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Extension 20090810: Support for automatic bridging COMMAND TO START RTPPROXY: rtpproxy -F -l 66.41.221.WAN/192.168.1.LAN -s unix:/var/run/rtpproxy.sock -d DBUG:LOG_LOCAL0 RTPPROXY DEBUG LOG: Nov 2 21:54:18 localhost rtpproxy[3408]: INFO:handle_command: new session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tag XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1 requested, type strong Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:create_twinlistener: can't bind to the IPv4 port 64918: Cannot assign requested address Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:handle_command: can't create listener Nov 2 21:54:18 localhost rtpproxy[3408]: INFO:handle_command: new session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tag XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1 requested, type strong Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:create_twinlistener: can't bind to the IPv4 port 64918: Cannot assign requested address Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:handle_command: can't create listener Nov 2 21:54:21 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found Nov 2 21:54:21 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found Nov 2 21:54:22 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found Nov 2 21:54:24 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found ####################Wireshark Packet Capture No. Time Source Destination Protocol Info 1 0.000000 192.168.1.LAN 204.235.61.9 NTP NTP client Frame 1 (90 bytes on wire, 90 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 204.235.61.9 (204.235.61.9) User Datagram Protocol, Src Port: ntp (123), Dst Port: ntp (123) Network Time Protocol No. Time Source Destination Protocol Info 2 0.069184 204.235.61.9 192.168.1.LAN NTP NTP server Frame 2 (90 bytes on wire, 90 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 204.235.61.9 (204.235.61.9), Dst: 192.168.1.LAN (192.168.1.LAN) User Datagram Protocol, Src Port: ntp (123), Dst Port: ntp (123) Network Time Protocol No. Time Source Destination Protocol Info 3 4.812402 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data, Frame 3 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1, Ack: 1, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 4 4.813081 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data Frame 4 (104 bytes on wire, 104 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1189, Ack: 1, Len: 38 [Reassembled TCP Segments (1189 bytes): #3(1151), #4(38)] Secure Socket Layer No. Time Source Destination Protocol Info 5 4.813109 192.168.1.LAN 66.87.78.CALLER TCP sip-tls > 49232 [ACK] Seq=1 Ack=1227 Win=1002 Len=0 TSV=1433086 TSER=34152821 Frame 5 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1, Ack: 1227, Len: 0 No. Time Source Destination Protocol Info 6 4.814100 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 6 (631 bytes on wire, 631 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1, Ack: 1227, Len: 565 Secure Socket Layer No. Time Source Destination Protocol Info 7 4.893013 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=1227 Ack=566 Win=2504 Len=0 TSV=34152854 TSER=1433086 Frame 7 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1227, Ack: 566, Len: 0 No. Time Source Destination Protocol Info 8 4.905470 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data, Application Data Frame 8 (556 bytes on wire, 556 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1227, Ack: 566, Len: 490 Secure Socket Layer No. Time Source Destination Protocol Info 9 4.923685 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data, Frame 9 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1717, Ack: 566, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 10 4.923755 192.168.1.LAN 66.87.78.CALLER TCP sip-tls > 49232 [ACK] Seq=566 Ack=2905 Win=1002 Len=0 TSV=1433196 TSER=34152854 Frame 10 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 566, Ack: 2905, Len: 0 No. Time Source Destination Protocol Info 11 4.996184 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data Frame 11 (280 bytes on wire, 280 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 2905, Ack: 566, Len: 214 [Reassembled TCP Segments (1365 bytes): #9(1151), #11(214)] Secure Socket Layer No. Time Source Destination Protocol Info 12 4.999143 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 12 (503 bytes on wire, 503 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 566, Ack: 3119, Len: 437 Secure Socket Layer No. Time Source Destination Protocol Info 13 5.002222 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 13 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 14 5.002329 192.168.1.LAN 66.87.76.CALLEE TLSv1 Application Data Frame 14 (1383 bytes on wire, 1383 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1, Ack: 1, Len: 1317 Secure Socket Layer No. Time Source Destination Protocol Info 15 5.103116 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=1003 Win=2539 Len=0 TSV=34152876 TSER=1433272 Frame 15 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 1003, Len: 0 No. Time Source Destination Protocol Info 16 5.700164 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Application Data Frame 16 (620 bytes on wire, 620 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 1, Ack: 1318, Len: 554 Secure Socket Layer No. Time Source Destination Protocol Info 17 5.700204 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=555 Win=998 Len=0 TSV=1433973 TSER=6923455 Frame 17 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 555, Len: 0 No. Time Source Destination Protocol Info 18 5.837983 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Application Data Frame 18 (812 bytes on wire, 812 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 555, Ack: 1318, Len: 746 Secure Socket Layer No. Time Source Destination Protocol Info 19 5.838020 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=1301 Win=998 Len=0 TSV=1434111 TSER=6923469 Frame 19 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 1301, Len: 0 No. Time Source Destination Protocol Info 20 5.838533 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 20 (695 bytes on wire, 695 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1003, Ack: 3119, Len: 629 Secure Socket Layer No. Time Source Destination Protocol Info 21 6.002208 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 21 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 22 6.002946 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=1632 Win=2574 Len=0 TSV=34152966 TSER=1434111 Frame 22 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 1632, Len: 0 No. Time Source Destination Protocol Info 23 7.001986 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 23 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 24 7.988054 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 24 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 1301, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 25 7.988112 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=2489 Win=998 Len=0 TSV=1436261 TSER=6923682 Frame 25 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 2489, Len: 0 No. Time Source Destination Protocol Info 26 7.996188 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 26 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 2489, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #24(1151), #26(70)] Secure Socket Layer No. Time Source Destination Protocol Info 27 7.996206 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=2559 Win=1002 Len=0 TSV=1436269 TSER=6923682 Frame 27 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 2559, Len: 0 No. Time Source Destination Protocol Info 28 7.997972 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 28 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1632, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 29 8.003228 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 29 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 30 8.174409 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=2757 Win=2619 Len=0 TSV=34153182 TSER=1436271 Frame 30 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 2757, Len: 0 No. Time Source Destination Protocol Info 31 8.515198 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 31 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 2559, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 32 8.515251 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=3747 Win=998 Len=0 TSV=1436788 TSER=6923734 Frame 32 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 3747, Len: 0 No. Time Source Destination Protocol Info 33 8.516100 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 33 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 3747, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #31(1151), #33(70)] Secure Socket Layer No. Time Source Destination Protocol Info 34 8.516116 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=3817 Win=1002 Len=0 TSV=1436789 TSER=6923734 Frame 34 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 3817, Len: 0 No. Time Source Destination Protocol Info 35 8.517743 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 35 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 2757, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 36 8.583042 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=3882 Win=2665 Len=0 TSV=34153224 TSER=1436790 Frame 36 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 3882, Len: 0 No. Time Source Destination Protocol Info 37 9.003201 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 37 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 38 9.621092 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 38 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 3817, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 39 9.621125 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=5005 Win=998 Len=0 TSV=1437894 TSER=6923837 Frame 39 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 5005, Len: 0 No. Time Source Destination Protocol Info 40 9.621136 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 40 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 5005, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #38(1151), #40(70)] Secure Socket Layer No. Time Source Destination Protocol Info 41 9.621143 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=5075 Win=997 Len=0 TSV=1437894 TSER=6923837 Frame 41 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 5075, Len: 0 No. Time Source Destination Protocol Info 42 9.622121 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 42 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 3882, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 43 9.684112 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=5007 Win=2710 Len=0 TSV=34153334 TSER=1437895 Frame 43 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 5007, Len: 0 No. Time Source Destination Protocol Info 44 10.003000 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 44 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 45 11.545138 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 45 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 5075, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 46 11.545187 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=6263 Win=998 Len=0 TSV=1439818 TSER=6924040 Frame 46 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 6263, Len: 0 No. Time Source Destination Protocol Info 47 11.545212 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 47 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 6263, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #45(1151), #47(70)] Secure Socket Layer No. Time Source Destination Protocol Info 48 11.545222 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=6333 Win=997 Len=0 TSV=1439818 TSER=6924040 Frame 48 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 6333, Len: 0 No. Time Source Destination Protocol Info 49 11.546755 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 49 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 5007, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 50 11.623085 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=6132 Win=2755 Len=0 TSV=34153528 TSER=1439819 Frame 50 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 6132, Len: 0 No. Time Source Destination Protocol Info 51 15.600662 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 51 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 6333, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 52 15.600719 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=7521 Win=998 Len=0 TSV=1443873 TSER=6924444 Frame 52 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 7521, Len: 0 No. Time Source Destination Protocol Info 53 15.601394 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 53 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 7521, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #51(1151), #53(70)] Secure Socket Layer No. Time Source Destination Protocol Info 54 15.601411 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=7591 Win=1002 Len=0 TSV=1443874 TSER=6924444 Frame 54 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 7591, Len: 0 No. Time Source Destination Protocol Info 55 15.601851 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 55 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 6132, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 56 15.666175 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=7257 Win=2800 Len=0 TSV=34153932 TSER=1443874 Frame 56 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 7257, Len: 0

2 1

Msilo: remove messages from db immediately after successful delivery
by Marc M. 05 Nov '13

05 Nov '13

Hi! I have a running msilo config. However, messages remain in the db after after successful delivery, and get removed at the next clean period. Unfortunatelly, this makes support cases dificult, as we do not see from the db which messages are delivered and which are not. My goal is is either: Remove the message from the db immediately after successful delivery (preferred) and/or be able to see the individual message delivery status. Can you help me, how should I proceed? Many thanks in advance! Marc

2 1

How to configure MySQL?
by Gus Monti 05 Nov '13

05 Nov '13

Hi, I can see that we are missing a lot of Sip messages in or db, so, checking Homer FAQ there is a way to compile MySql with Insert Delayed for partitioning table. The problem is that I don't understand how to recompile MySql and make the necessary changes. The URL is https://code.google.com/p/homer/wiki/FAQ#Q:_I_have_a_lot_of_SIP_traffic_an d_some_SIP_messages_are_missing The question is, can you send me a step by step how to do this? I really appreciate your help. I have CentOS 6.4: [root@Homerdb02 ~]# kamailio -v version: kamailio 4.1.0-dev9 (x86_64/linux) f8f3d3 [root@Homerdb02 ~]# cat /etc/redhat-release Red Hat Enterprise Linux Server release 6.4 (Santiago) [root@Homerdb02 ~]# uname -a Linux Homerdb02 2.6.32-358.18.1.el6.x86_64 #1 SMP Fri Aug 2 17:04:38 EDT 2013 x86_64 x86_64 x86_64 GNU/Linux [root@Homerdb02 ~]# mysql -V mysql Ver 14.14 Distrib 5.5.34, for Linux (x86_64) using readline 5.1 Thanks, Gus

2 1

Event route sl:local-response is not executed in kamailio 4.1
by Sergey Okhapkin 04 Nov '13

04 Nov '13

I just upgraded one server to latest git (version: kamailio 4.1.0-pre0 (x86_64/linux) 26fa46-dirty), sl:local-response route is no longer executed. I have in kamailio.cfg event_route[sl:local-response] { xlog("L_INFO","Local response\n$mb\n"); } No local responses are logged in kamailio log file. It was working fine in kamailio 4.1.0-dev7 (x86_64/linux) b2a536-dirty

2 2

Status : 401 Unauthorized (0 bindings) Reason
by Owais ul Haq 04 Nov '13

04 Nov '13

Hello, I am using kamailio server. Problem is when my soft phone tries to connect to Kamailio server, it connects in second attempt. First it gets a sip reply saying "Status : 401 Unauthorized (0 bindings)". I checked this using Wire Shark. Can some one please help me in finding the reason for this error message found all the time. Thanks Best Regards

2 1

Reason for 401 Unauthorized message
by Owais ul Haq 04 Nov '13

04 Nov '13

Hello, I am using kamailio server. Problem is when my soft phone tries to connect to Kamailio server, it connects in second attempt. First it gets a sip reply saying "Status : 401 Unauthorized (0 bindings)". I checked this using Wire Shark. Can some one please help me in finding the reason for this error message found all the time. Thanks Best Regards

2 1

Limit the nr. of transaction retries
by Marc M. 04 Nov '13

04 Nov '13

Hi! I Have a questions concerning t_realy() and failure route. If the t_realy() is not successful, it makes 12 re-tries before failure route is called. Can you help me, how can I lower the number of retries before move to failure route? My goal is to make 3 re-tries and then call failure route. (I gues it is a tm param?) Thansk you for your help in advance! regards Marc

2 1

Kamailio callee timeout problem
by kamailio＠AaronLux.com 04 Nov '13

04 Nov '13

I'm pretty sure I'm just missing something very simple here like a port forward. Let me know if you have any ideas! This setup completely works when both csipsiimple clients are inside my LAN and kamailio is configured with alias=192.168.1.LAN. The problem begins when I connect both clients from the WAN and configure kamailio with alias=66.41.221.WAN. The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out. Summary:: The callee and caller both establish 2-way audio and zrtp keys. TLS is working. Both SIP clients are using the OSTN wizard in csipsimple. Please review my Debug logs: http://pastebin.com/9Rw7zSQ0

2 1

Regex question
by Oliver Roth 04 Nov '13

04 Nov '13

Hi all Maybe a very simple question - but I cannot see the solution - I am more or less newbie ;) I need to do some string operations in kamailio.cfg. How can I get the cli from the following string: sip:+41523940347@195.216.67.103;user=phone I only would need +41523940347 in a variable Something like $avp(s:myCli) = ^sip:\+(\d{11})@.{1,40}$ Any idea? Thanks in advance Oli

4 4

Re: [SR-Users] Capturing-200 Okey(with SDP) from cfg (Daniel Grotti)
by Surendra 04 Nov '13

04 Nov '13

Hi Daniel, Issue got solved. I am just sharing it. My requirement is I have to parse the SDP from cfg which is coming in 200 OKEY(Initiated by UAC once the called party answers the call). On reply route we can capture the responses for those requests which we set t_on_reply();(externally we have to call for each request). But reply route we can capture all replies. So that we can parse our required headers or msg body. Regards Surendra ******************************

1 0

TLS: support TLS v1.2 ?
by Marc M. 03 Nov '13

03 Nov '13

Hi! I have question concerning TLS module. I would like to enable TLS 1.2 protocol. As I understand the TLS module is compiled with openssl, which as far as i know has TLS 1.2 support since openssl 1.0.1. I am wondering if there is way to enable TLS 1.2 on the TLS module? Many thanks Marc

2 1

Incoming call problem
by Jason Sia 02 Nov '13

02 Nov '13

Hi, I have kamailio server and a freeswitch server during a call the kamailio passes the call to freeswitch to play a media and back to kamailio for routing to the destination, this works fine however for samsung devices (any ROM using linphone). On other device brands, it does not work it always returns USER_BUSY or NUMBER_NOT_FOUND. After investigation, I found out that if the transport is UDP then it works on all phones, the problem happens if I use TCP as the transport. Anyone had experienced on this? Thanks, Jason

3 4

Re: [SR-Users] xHttp_PI Module
by Ovidiu Sas 02 Nov '13

02 Nov '13

Please post your questions to the mailing list instead of sending private e-mails. The subscriber provisioning via xhttp is supported only for plaintext passwords. Set modparam("auth_db", "calculate_ha1", 1) and it will work fine. Regards, Ovidiu Sas On Sat, Nov 2, 2013 at 12:22 PM, Abdul Hakeem <alhakeem(a)gmail.com> wrote: > Hello, > > I just want to enquire if the subscriber full provisioning is ready. > Best regards, > Abdul Hakeem > > -- VoIP Embedded, Inc. http://www.voipembedded.com

1 0

dispatcher: how to set the transport protocol
by Karimi-Cherkandi, Bizhan 01 Nov '13

01 Nov '13

kamailio dispatcher module in my environment uses only UDP I have not been able to find how I can configure kamailio dispatcher module to use TCP Would appreciate any help Bijan

3 6

Kamailio - RTPProxy - Asymmetric RTP
by Lucas Girard 01 Nov '13

01 Nov '13

Hi all, I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is used. I have the system running, lines registered in Kamailio are able to call to several destinations, but I have a problem with one provider that use Asymmetric RTP (others providers use Symmetric RTP and I dont have any issues). A -----> Kamailio & RTPProxy ----------> Trunk -------> Provider (Asymmetric RTP) -------> B A is calling B. The problem is that line in kamailio A can listen to B, but B can not listen what A says. I have captured the sip dialog and rtp traffic and I can see that: Kamailio tells to provider that is using the port 52388 (media port in SDP). Providers tells Kamailio that is using the port 5394 (media port in SDP). RTP traffic from Providers to kamailio goes from port 5392 to 52388. RTP traffic from Kamailio to Provider goes from port 52388 to 5392. I have not found an RFC regarding Asymmetric RTP. The question are, why RTP is sent from Kamailio (rtpproxy) to Provider to destination port 5392? It should use 5394 as SDP indicate? or, port 5392 is learned when RTP traffic was received from 5392 to 52388? # kamailio -V version: kamailio 3.3.0 (i386/linux) 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, USE_FUTEX, 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, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:17:07 Jun 18 2012 with gcc 4.4.5 # rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications I hope to be clear with this explanation. Regards, Lucas Girard

3 7

msilo - use case: registered but not reachable
by Marc M. 01 Nov '13

01 Nov '13

Hi! I have a simple msilo setup which is working fine: -If the user is registered, the messages are delivered right away -If the user is not registered, the messages are stored in msilo and got deliverd at the next register Here is the relevant part of the cfg: --------------- if (is_method("MESSAGE")) { log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n"); route(MSILO_MESSAGE); } . . . route[MSILO_MESSAGE]{ if (lookup("location")){ t_relay(); }else{ m_store("$ru"); sl_send_reply("200", "OK"); } exit; } . . . route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); # MSILO - dumping user's offline messages m_dump(); exit; } } ------------------- Thats working fine. Now, we have a use case where a user is registered, but got terminated in between abd not reachable. So the user is in the userloc table, but not reachable. In this case ,since the user is still in userloc, the t_relay() will be executed, and it drops a 408 (timeout) code which is fine. However, the process end here. The message is then dropped. What i would like to achieve is to make sure that t_relay() was executed with 200, otherwise store the message and deliver it at the next register. I was trying this: ------------ ##storing offline messages (new) route[MSILO_MESSAGE]{ if (lookup("location")){ if (!t_relay()) { m_store("$ru"); sl_send_reply("200", "OK"); } exit; } else{ m_store("$ru"); sl_send_reply("200", "OK"); } exit; } -------- but i have still the same behaviour. Can you help me, what do I wrong? Many thnaks Marc. M

2 1

Re: [SR-Users] Capturing-200 Okey(with SDP) from cfg
by Daniel Grotti 01 Nov '13

01 Nov '13

Hi, what do you mean with "capture from cfg file" ? Do you want to log 200OK message in kamailio log file ? Daniel On Friday, November 1, 2013 08:36 CET, "Surendra" <surendra.pullaiah(a)plintron.com> wrote: > Hi all, > > > > I have a requirement , i.e. I need to capture 200 ok(with > SDP) from cfg file. This 200 okey will be getting once the called party > answers the call. > > Can anyone help me to get out of this. > > > > Regards > > Surendra > > > > >

1 0

Capturing-200 Okey(with SDP) from cfg
by Surendra 01 Nov '13

01 Nov '13

Hi all, I have a requirement , i.e. I need to capture 200 ok(with SDP) from cfg file. This 200 okey will be getting once the called party answers the call. Can anyone help me to get out of this. Regards Surendra

1 0

Help : Kamailio RTP Proxy Issue...
by Mahmoud Ramadan Ali 30 Oct '13

30 Oct '13

Hello , This is the first time to put a question on sr-users mailing list and i hope to help me to solve my issue...i'm following this post http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html to setup RTP Proxy for kamailio 4 but it's NOT working as predicted and i'm receiving this error message on asterisk... [2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)", ...): Name or service not known [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 192.168.1.80192.168.1.80' [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)... The problem is NOT at the asterisk side but at the proxy side...when kamailio rewrites the SDP message it adds the public IP twice in the "C" filed of the SDP message instead of only one time like this 'IN IP4 192.168.1.80192.168.1.80' this is because the scripte is calling rtpproxy_manage() in the routes TOASTERISK /FROMASTERISK and in MANAGE_BRANCH / NATMANAGE. It should be either or,not both....but i don't know how to do this since i don't have any scripting skills so if anyone can modify the config file in the attachment i will be thankful... Best Regards' Mahmoud Ramadan Ali

2 2

Hardware requirements/recommandations
by Igor Potjevlesch 30 Oct '13

30 Oct '13

Hello, I'm looking for the best practises for setting Kamailio and the associated hardware requirements. Kamailio must deliver registrar and routing functions for 500k aliases, 5k registered accounts and ~10k routes set in carrierroute module or similar. My worries mainly concern DB integration. Is it better to setup Kamailio DB on the same server (no network issue, etc.) or on another server? Your opinion/feedback experience would be appreciated. Thanks! Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 7

Upgrade v4.0.x to 4.1.0 guidelines
by Daniel-Constantin Mierla 30 Oct '13

30 Oct '13

Hello, this is a call for developers and community users to update the page collecting upgrade guidelines from current stable (4.0.x) to upcoming major release (4.1.0): - https://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0 If you are aware of a change in the behaviour for some parameters, functions or statements, list them there. I added the sql statements for MySQL that will allow to get to the new database structure. Would be good to have the same for other popular sql servers (maybe postgres, sqlite ...). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Inser Delayed in MySQL
by Gus Monti 29 Oct '13

29 Oct '13

Hi, I can see that we are missing a lot of Sip messages in or db, so, checking Homer FAQ there is a way to compile MySql with Insert Delayed for partitioning table. The problem is that I don't understand how to recompile MySql and make the necessary changes. The URL is https://code.google.com/p/homer/wiki/FAQ#Q:_I_have_a_lot_of_SIP_traffic_an d_some_SIP_messages_are_missing The question is, can you send me a step by step how to do this? I really appreciate your help. I have CentOS 6.4: [root@Homerdb02 ~]# kamailio -v version: kamailio 4.1.0-dev9 (x86_64/linux) f8f3d3 [root@Homerdb02 ~]# cat /etc/redhat-release Red Hat Enterprise Linux Server release 6.4 (Santiago) [root@Homerdb02 ~]# uname -a Linux Homerdb02 2.6.32-358.18.1.el6.x86_64 #1 SMP Fri Aug 2 17:04:38 EDT 2013 x86_64 x86_64 x86_64 GNU/Linux [root@Homerdb02 ~]# mysql -V mysql Ver 14.14 Distrib 5.5.34, for Linux (x86_64) using readline 5.1 Thanks, Gus

2 2

Kamailio Tech Admin Group
by Daniel-Constantin Mierla 29 Oct '13

29 Oct '13

Hello, being discussed during last Devel IRC Meeting, we are planing to build a Kamailio Project Technical Administration Group: https://www.kamailio.org/wiki/devel/irc-meetings/2013blog#technical_adminis… Its goal is to get a bunch of people that volunteer to do administration tasks for the project, such as: - helping with releases (e.g., patch backports, packaging, uploading files for download, etc) - doing sysadmin tasks for our servers (e.g., performing upgrades to wiki, web site, etc) - preparing technical decisions and doing them (e.g., what applications to use to make operations easier, cloning git repository to github, ...) From the devel meeting, so far we have Victor Seva, Fred Posner, Peter Dunkley and Olle Johansson. Existing people doing admin tasks will probably stay in (if they don't opt out): me, Elena-Ramona Modroiu, Henning Westerholt (owner of devel.kamailio.org hardware), Jan Janak (owner of sip-router.org hardware), Jesus Rodriguez and Oriol Capsada (owners of kamailio.org hardware). Requirements for candidates and other details: - volunteer to do the work, it is not a paid job - an existing record of activity within the project is a plus (e.g., developer, active mailing list member) - reply to the lists detailing where and how you can help - possibility to spend 1-4 hours a week for project administration (more is welcome, sometime is not necessary at all) Rewards: - you will be listed as part of project administration on the website - get to interact more with the project and the nice guys around it ;-) - more spam - admin list address will be public and the list open so everyone can send in case of critical situations (content/archive will be kept private) Note that we will try to build a group of an adequate size, thus not everyone willing to participate may get in (at least on the first phase). One criteria is to have skills that complement existing team knowledge. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

5 5

ACK not sending to the invite to header but to the contact.
by anfecora 29 Oct '13

29 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. any help is apreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. . U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT') U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. . U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9. From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .

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Dialog callback after restart
by Kristian F. Høgh 29 Oct '13

29 Oct '13

Hi, After restarting kamailio, pua_dialoginfo doesn't get updated when active dialogs change status. I see the dialogs are read from the database, but when they receive eg. BYE the callback are not called. I use db_mode 0 in pua, but tried mode 2 with no luck. Should pua_dialoginfo register DLGCB_LOADED? Regards, Kristian Høgh.

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Re: [SR-Users] Kamailio network edge for registration and rtp pass-through
by JR Richardson 29 Oct '13

29 Oct '13

Date: Fri, 25 Oct 2013 17:48:13 -0400 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: SR-Users <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio network edge for registration and rtp pass-through Message-ID: <6D231DE3-5BED-4601-A097-5A79CEF452F0(a)palner.com> Content-Type: text/plain; charset=us-ascii Do you need the registration be local to the asterisk? I would have all the asterisks send calls to the Kamailio. You can have a lookup on endpoint outbound to decide which asterisk should handle the outbound call for that did. Also a lookup for incoming DIDs, etc. ---Fred > On Oct 25, 2013, at 5:43 PM, Jr Richardson <jmr.richardson(a)gmail.com> wrote: > > Hi All, > > Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. > > I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. > > phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 > phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 > phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 > > I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. > > Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? > > So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? > > A point in the right direction? > > Thanks. > > JR I would like kamailio to be transparent in the process, i.e. phones register locally to their respective PBX. Thanks. JR

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Fwd: Presence Proxy
by Jan Gaida 29 Oct '13

29 Oct '13

Hello, I'd like to explain my scenario and the module I'd like to develop. I am hoping for comments whether there are already modules like that or which modules' API I could use to make the development easier. Scenario: I have two domains (A and B) with a Kamailio server in each of them configured as presence servers. The agents from both domains want to subscribe to few resources from domain A. Idea: In order to reduce traffic between the domains, the presence server from domain B shall handle the subscriptions internally and use an internal virtual presence-user-agent to subscribe to the resources in domain A. SIP Messages: Subscription: 1) Subscribe resource.domain.a, from each user agent ua_n.domain.b 2) 202 OK from kamailio.domain.b 3) If not already subscribed, Subscribe resource.domain.a from kamailio.domain.b with virtual user-agent 4) 202 OK from kamailio.domain.a (Repeat for each subscriber) (Subscription has to be refreshed by an internal timer) Events: 1) Publish resource.domain.a to kamailio.domain.a 2) 200 OK from kamailio.domain.a 3) Notify resource.domain.a to kamailio.domain.b 4) 200 OK from kamailio.domain.b 5) Notify resource.domain.a to each user agent ua_n.domain.b 6) 200 OK from each ua_n.domain.b (Internally, Kamailio of domain B has to forward the incoming Notify to all locally subscribed user-agents) I tried using the PUA module but it does not fit all my needs. Now I think I will have to develop my own module in order to achieve everything. Any ideas? Kind regards Jan Gaida PS: Sorry for mailing to both mail-lists. But I was not sure which list was the correct for this topic. -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

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Re: [SR-Users] Pickup group implementation in Kamailio
by Daniel-Constantin Mierla 29 Oct '13

29 Oct '13

Hello, for pickup group look at pua_/presence_dialoginfo or sca modules. Cheers, Daniel On 10/23/13 11:52 AM, John Murray wrote: > > Hi, > > I am also looking for a 'group pickup' function with Kamailio. > > Once I have the information on the call I want to get how would I > 'grab' the call. > > My calls are being processed with the tm module. > > I imagine I send an INVITE with the 'replaces' header but what > Kamailio module would process that and what would the config look like? > > Many thanks > > John > > >/ Hello,/ > > >// > > >/this is a custom implementation, only for phones that have programmable/ > > >/buttons. It is not really recommended way to do it./ > > >// > > >/Better rely on standard specs out there and use presence extensions for/ > > >/such feature -- in kamailio you would need to use modules such as > dialog,/ > > >/presence, pua, presence_dialoginfo and pua_dialoginfo./ > > >// > > >/Cheers,/ > > >/Daniel/ > > >// > > >// > > >/On 5/14/12 1:15 AM, M.C. wrote:/ > > >// > > >/Hi all,/ > > >/I want to implement a "group pickup" (asterisk like) with Kamailio and/ > > >/Snom phones./ > > >/I found in the changelog this comment:/ > > >/"- an useful application - call pickup:/ > > >/ - INVITE comes in, callee is in a pickup group/ > > >/ - store callid, cseq, etc in database or notify the other phones in/ > > >// > > >/ the group (easy to do for snom phones and extra programmable/ > > >/ buttons) via uac_req_send()/ > > >/ - send a remote control command telling which call to pickup/ > > >/ (callid, cseq) and where to redirect. In config, call/ > > >/ t_cancel_callid(). in failure route for INVITE catch the > cancelled/ > > >/ transaction (flag is set), get the new destination (e.g., > stored in/ > > >/ htable) and forward the invite there""/ > > >// > > >/regarding new functions in tmx module, but I have found no examples > in kamailio documentation./ > > >// > > >/Is this the way to implement a group pickup in kamailio?/ > > > > _______________________________________________ > 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 Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

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ALIAS_DB module resolving to multiple user@domain does not work when use_domain is set to 1
by Vassilis Radis 29 Oct '13

29 Oct '13

The documentation of the append_brances(int) function in ALIAS _DB module says: "If the alias resolves to many SIP IDs, the first is replacing the R-URI, the rest are added as branches." However this only works if not using the domain for resolving, i.e. use_domain is set to 0. This happens because there is a unique constraint on the dbaliases table for the columns (alias_username, alias_domain). Would it hurt to drop the constraint so that I can have multiple branches when resolving username AND domain? Any other suggestions/comments? Thanks.

2 1

How to add debugger module in kamailio.cfg
by Owais ul Haq 29 Oct '13

29 Oct '13

Hello, I want to add "dubugger module" in kamailio. As I want to know which part of kamailio.cfg is responsible for reply of some of incoming sip messages. Can some one please help me in this regard. Regards, Kamailio kid.

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Help with 200k responses to the contact header not the record-route
by anfecora 29 Oct '13

29 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok properly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. I am sorry i sent a too long message before i will try skim it a bit. any help is appreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0.

2 1

Regarding Exporting a function
by premchandiran.marimuthu 29 Oct '13

29 Oct '13

Hi All, I was able to export a function to cfg file with int return type but when i try to export a function with char/char-ptr as return type , in cfg file i was not able to print/use the characters which i returned in the exported function. can i get a clue/tip to overcome above said issue?. Regards, Prem Chandiran M -- Open WebMail Project (http://openwebmail.org)

2 1

Presence issues
by Wingsravi R 29 Oct '13

29 Oct '13

Hi kamailio'ns I am working on Kamailio V 4.0.3, installed from Ubuntu 12.04(precise) repository. My SIP clients(jitsi) are registering to kamailio sever successfully but their presence is not sharing between the clients. I am frequently logging following presence related errors in syslog while trying to register SIP clients to kamailio server: Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15274]: INFO: pua_usrloc [ul_publish.c:221]: ul_publish(): not marked for publish Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15273]: WARNING: presence [publish.c:481]: handle_publish(): Missing or unsupported event header field value Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15273]: ERROR: presence [publish.c:484]: handle_publish(): event=[reg] what could be the wrong ? How can i reslove this issue ? please find in the attachment is my kamailio config file. (suggest me if any changes to be made to this configuration) Any help will greatly appreciate. Regards, Ravi

2 3

kamailio 3.3.2 db_mysql.so problem after mysql upgrade to 5.6
by Uri Shacked 28 Oct '13

28 Oct '13

Hi, I upgraded mysql from 5.5 to 5.6. After the upgrade kamailio fails to start and throws the following error: "..... /usr/lib64/libmysqlclient.so.18: version `libmysqlclient_16` not found....." Any ideas what to do? Thanks, Uri

2 2

media between 2 clients behind separate NAT
by Vassilis Radis 28 Oct '13

28 Oct '13

Hello, When I have 2 clients using a kamailio proxy, and both of the clients are behind their own NAT, then my only options for relaying media between them is using some kind of intermediate rtp proxy or STUN etc? Proxy is on a public IP and essentially I am asking if I can avoid setting up stuff to proxy the media or mess with STUN, by using some kind of request response mangling. i.e. have both clients's sdp address/port changed to their public facing ip/port and then each client can send an initial packet to the other end. Those initial packets will be blocked by the other receiver router, but they will open a nat hole so that next package from the peer will pass through. Is that feasible? Thanks.

2 1

How can I use a module from the devel branch in a stable varsion installation
by Vassilis Radis 27 Oct '13

27 Oct '13

Hello, I would like to use rtpproxy-ng module from the devel branch in an 4.0.4 installation. I have succesfully compiled the mediaproxy-ng kernel modules. I would like to know if there is a way to compile the rtpproxy-ng module in the devel branch in such a way that I dont get the following error "kamailio: ERROR: <core> [sr_module.c:422]: version_control(): ERROR: module version mismatch for /usr/local/lib/kamailio/modules/rtpproxy-ng.so; core: kamailio 4.0.4 (i386/linux); module: kamailio 4.1.0-pre0 (i386/linux)" Thanks.

2 2

Issue with failure_route on kamailio 4.0
by mayamatakeshi 27 Oct '13

27 Oct '13

Hello, I am trying to upgrade from kamailio 1.4 to 4.0. I am having an issue with call forwarding: I send the call to a subscriber (location) and if it doesn't answer, there is a failure_route to do the forward. It is something like this: failure_route[1] { $ru = "sip:0313241342@192.168.2.2" append_branch(); t_relay(); } However, the INVITE is sent to both the subscriber location and the new branch. So, is it now necessary to somehow to "drop" the original branch? If so, how do I do that? Regards, Takeshi

2 2

Usrloc errors
by Wingsravi R 26 Oct '13

26 Oct '13

Hi kamailio'ns I am working on MSRP module of Kamailio serverV 4.0.3, which is installed from Ubuntu 12.04(precise) repository. When i am trying to register SIP clients(Blink) to kamailio server, frequently logging following user location related errors in syslog, SIP clients are registering to kamailio sever successfully, but still logging these errors and client presence is not visible to other client. Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Duplicate entry 'uloc-526a7929-3bab-91' for key 'ruid_idx' Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: usrloc [ucontact.c:625]: db_insert_ucontact(): inserting contact in db failed Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: usrloc [urecord.c:368]: wb_timer(): updating contact in db failed (aor: 1112(a)kamailio.test) Below attachment is my Kamailio config file (please suggest me if any changes to be made). Anybody please help me in resolving the issues PS: With the same script when i used to register the clients (jitsi) to kamailio i am logging presence related errors (which i posted too in the mailing list with the subject as : Presence issues) Regards, Ravi

2 2

Kamailio network edge for registration and rtp pass-through
by Jr Richardson 25 Oct '13

25 Oct '13

Hi All, Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? A point in the right direction? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Help With kamailio not responding ack propperly.
by anfecora 25 Oct '13

25 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok properly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. any help is appreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. . U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT') U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. . U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9. From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .

2 3

Invite Fragmented
by Gus Monti 25 Oct '13

25 Oct '13

Hi, There are often times where key packets are missing from homer traces. This seems to be caused by packet fragmentation. It is most readily evidenced by the trace showing an Invite entering the SBC from the outside and then no invite on the inside being seen but the 100 trying is seen on both sides. Any Idea why? Thanks, Gus

3 2

RTP Proxy issue on Kamailio 4
by Mahmoud Ramadan Ali 25 Oct '13

25 Oct '13

*Hello , This is the first time to put a question on *sr-users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> mailing list * and i hope to help me to solve my issue...i'm following this post http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html to setup RTP Proxy for kamailio 4 but it's NOT working as predicted and i'm receiving this error message on asterisk...* [2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)", ...): Name or service not known [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 192.168.1.80192.168.1.80' [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)... The problem is NOT at the asterisk side but at the proxy side...when kamailio rewrites the SDP message it adds the public IP twice in the "C" filed of the SDP message instead of only one time like this 'IN IP4 192.168.1.80192.168.1.80' but i don't know why it's doing this...? if u have any idea or recommendations plz tell me cuz i'm new to kamailio...waiting ur reply and thanks in advance... also here's my config file if u want to take a look at it... #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v4.0 - 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 # # *** 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 ####### 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://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://root:passw0rd@192.168.1.10/mya2billing" #!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=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=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 mhomed=1 ####### 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 = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.10" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.1.40" 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/local/lib/kamailio/modules_k/:/usr/local/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" loadmodule "dispatcher.so" loadmodule "ipops.so" loadmodule "sqlops.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!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") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!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", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "secret") 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) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif # ------- Load-balancer params ------ modparam("dispatcher", "db_url","mysql://kamailio:kamailiorw@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:Kamailio-Proxy@192.168.1.40") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_threshhold", 3) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") # ----- SQLops params ----- modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost /kamailio") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))){ return; } route(NATMANAGE); #!ifdef WITH_ASTERISK route(REGFWD); #!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 #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); 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 #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "cc_sip_buddies", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { if (is_method("INVITE")){ sql_query("ca", "select destination from dispatcher where destination like '%$dd%'","ra"); if($dbr(ra=>rows)>0){ $avp(duip)=$(du{s.select,-2,:}); if (is_ip_rfc1918("$avp(duip)")) { xlog("L_INFO", "Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); #rtpproxy_manage("crwie","192.168.1.40"); rtpproxy_manage("rwie"); } } else if(ds_is_from_list()){ if (is_ip_rfc1918("$si")) { xlog("L_INFO", " Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); #rtpproxy_manage("crwei","192.168.1.80"); rtpproxy_manage("rwei"); } }else if(!ds_is_from_list()){ rtpproxy_manage("rwie"); } } } # 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 = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if(ds_is_from_list()){ rtpproxy_manage("cawie"); xlog("L_INFO","[$fU@$si:$sp]{$rm} Call from Media-Server Cluster\n"); return 1; } return -1; } # Send to Asterisk route[TOASTERISK] { ds_mark_dst("IP"); if(!ds_select_dst("1", "9")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n"); exit; } xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); rtpproxy_manage("cawei"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[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) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($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 2

Debugging Logs Location
by Owais ul Haq 25 Oct '13

25 Oct '13

Hi all, I have loaded debugger module in my kamailio server and set "cfgparameter" to 1. Can some one please tell me where can I now find logs of this cfg trace. Any help will be appreciated. Thanks. Regards, M. Owais ul Haq

2 1

Audio Codec option in Siremis 4.0
by Adarsh Pillai 25 Oct '13

25 Oct '13

Hi Where do i find the option for enabling and disabling audio codecs in Siremis 4.0. Regards Adarsh Pillai

2 1

MSRP module - Ubuntu repo
by Wingsravi R 24 Oct '13

24 Oct '13

Dear Kamailio'ns, I have Installed kamailio V 4.0.3 from Ubuntu 12.04 (precise) repository and running successfully. now i have to work around MSRP relay module of kamailio sever to achieve file transfer between two SIP clients. so, i just tried to install MSRP module, but i get logged as below: /$ sudo apt-get install kamailio-msrp-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio-msrp-modules Is MSRP module is included in Ubuntu repository? how can i install it ? Thank you and regards,

2 6

proxy is not forwarding 180 ringing reply
by Juha Heinanen 24 Oct '13

24 Oct '13

i have been wondering why my proxy is not forwarding 180 ringing reply to caller that it receives from callee. invite was forwarded to callee by t_relay(). proxy does not print any error messages to syslog. the only thing i get is the message i have in onreply_route: onreply_route [REPLY] { xlog("L_INFO", "Received reply\n"); return; } level 3 debug of 180 reply processing is below. does it tell anything to tm gurus that would explain why kamailio does nothing to the reply? 180 reply itself looks like this: Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Record-Route: <sip:192.98.102.30;transport=tcp;r2=on;lr> Record-Route: <sip:192.98.102.30;r2=on;lr> Via: SIP/2.0/TCP 192.98.102.30;branch=z9hG4bK84cf.498ea35cdde08fdfad989b 9212a2a9a5.0 Via: SIP/2.0/UDP 192.98.102.31:55419;branch=z9hG4bKd6e2f0db78e2df94;rpor t=55419 To: <sip:test@test.tutpro.com>;tag=8e143c1b780cfb1b From: <sip:jh@test.tutpro.com>;tag=630450233fa9bc92 Call-ID: d0b328ba01bcc483 CSeq: 22643 INVITE Server: baresip v0.4.6 (i586/linux) Contact: <sip:0x94fa6e8@192.98.102.30:49240;transport=tcp> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,NOTIFY,SUBSCRIBE,INFO Content-Length: 0 -- juha Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [tcp_read.c:1510]: handle_io(): received n=4 con=0xb4252638, fd=15 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [tcp_read.c:1317]: tcp_read_req(): tcp_read_req: content-length= 0 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <180> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Ringing> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK84cf.498ea35cdde08fdfad989b9212a2a9a5.0>; state=16 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffff Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKd6e2f0db78e2df94>; state=6 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <55419>; state=16 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=62 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=8e143c1b780cfb1b Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, state=29 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:test@test.tutpro.com] Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:test@test.tutpro.com>] Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <22643> <INVITE> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 64584 label 0 branch 0 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb4256d88)! Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb4256d88, callback type 2, id 0 entered Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: acc [acc_logic.c:538]: tmcb_func(): acc callback called for t(0xb4256d88) event type 2, reply code 180 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=5 T end=0xb4256d88 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_reply.c:2258]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: INFO: Received reply Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x82ca540, 15, 2, 0xb4252638), fd_no=1

4 12

clearing pua table
by SIP Guru 24 Oct '13

24 Oct '13

Hi Kristian, thank you. I'll check this. I think in some cases we are using even a higher lifetime. Could you please send me a patch with your changes for the pua_dialoginfo module in case we run into the same issue. Toni On Tuesday 22 October 2013 13:19:01 Kristian Høgh wrote: >Hi, > >I think I've got the same issue. >At call creation we set dialog lifetime to 6 hours. >pua_dialoginfo uses that value and send PUBLISH for each hour, even after the dialog terminates. >After 6 hours the entry is deleted. > >To prevent the updates, I've made a small change to pua_dialoginfo module, which set pua-dialog timeout to 300 sec. when the dialog terminates. >pua_dialoginfo then sends PUBLISH, which updates presence. > >Regards, >Kristian Høgh > > >On Monday 21 October 2013 15:21:16 SIP Guru wrote: >>/ Hi, />>/ />>/ expired entries in pua table won't be deleted. /<>/ And kamailio keep sending PUBLISH's for expired entries. />>/ />>/ Am I missing something ? />>/ />>/ Toni />>/ />>/ />>/ />>/ _______________________________________________ />>/ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list />>/ sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> />>/ http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users />

3 3

Pickup group implementation in Kamailio
by John Murray 23 Oct '13

23 Oct '13

Hi, I am also looking for a 'group pickup' function with Kamailio. Once I have the information on the call I want to get how would I 'grab' the call. My calls are being processed with the tm module. I imagine I send an INVITE with the 'replaces' header but what Kamailio module would process that and what would the config look like? Many thanks John > Hello, > > this is a custom implementation, only for phones that have programmable > buttons. It is not really recommended way to do it. > > Better rely on standard specs out there and use presence extensions for > such feature -- in kamailio you would need to use modules such as dialog, > presence, pua, presence_dialoginfo and pua_dialoginfo. > > Cheers, > Daniel > > > On 5/14/12 1:15 AM, M.C. wrote: > > Hi all, > I want to implement a "group pickup" (asterisk like) with Kamailio and > Snom phones. > I found in the changelog this comment: > "- an useful application - call pickup: > - INVITE comes in, callee is in a pickup group > - store callid, cseq, etc in database or notify the other phones in > > the group (easy to do for snom phones and extra programmable > buttons) via uac_req_send() > - send a remote control command telling which call to pickup > (callid, cseq) and where to redirect. In config, call > t_cancel_callid(). in failure route for INVITE catch the cancelled > transaction (flag is set), get the new destination (e.g., stored in > htable) and forward the invite there"" > > regarding new functions in tmx module, but I have found no examples in kamailio documentation. > > Is this the way to implement a group pickup in kamailio?

1 0

Kamailio MSRP relay problem
by mohammed alyaseen 23 Oct '13

23 Oct '13

Hi all... I am trying to do file transfer between IMS clients (web-based WebRTC and android RCS native client). The fisrt test scenario is browser-to-browser (WebRTC) using JSSIP stack. The second scenario is RCS-to-Broswer or Browser-to-RCS. My testbed consists of OpenIMS and OverSIP for WebRTC clients and Kamailio MSRP Relay for file transfer For file transfer I am using the msrp-crocodile library... The scenario browser-to-browser works so far without problems. But in the second scenario I have the following problems: * When I try to transfer a file from the RCS client to WebRTC client , kamailio sends the message "error ""481", "Session-does-not-exist", however the file is sent, when I click send for the second time... * When I try to send a file from WebRTC client to RCS client, WebRTC client sends all file chunks to kamailio, kamailio receives the chunks but it doesn't forward them to the RCS client, in the console I see some errors pointed to the configuration file, however I don't know what exactly the problem is. In the attachment please find the logs and the configuration file of the kamailio msrp relay... Can any one help me to solve this problem or give me useful hints? Thanks a lot in advance.... Medo

2 3

why kamcmd tls.reload is not safe
by Ding Ma 22 Oct '13

22 Oct '13

In the current Kamailio TLS module document, there is a statement about tls.reload being unsafe. But the only way to periodically update CRL without restarting Kamailio is to use tls.reload. In our test with tls.reload for CRL, it seems Kamailio would crash after about 100 times of tls.reload in 5/6 hours. The core dump indicates memory access violation, signal 11. We compiled Kamailio with openssl 1.0.0-fips. Would appreciate some insights on tls.reload and ideas to fix the crash issue. Thanks,

3 4

Dialog profiles using $fU and $rU
by Daniel W. Graham 22 Oct '13

22 Oct '13

Having an odd issue with dialog profiles, if I try to set/check profile based on $fU or $rU call is always rejected due to call limit, but $ru or $fu work without an issue. Example using $fU- $avp(s:checkuser) = $fU; set_dlg_profile("quota","$avp(s:checkuser)"); dlg_manage(); get_profile_size("quota","$avp(s:checkuser)","$avp(s:calls)"); avp_db_load("$avp(s:checkuser)/username", "$avp(s:callquota)"); if ($avp(s:calls) <= $avp(s:callquota)) ....... If I change checkuser to $avp(s:checkuser) = $fu; it returns the correct result. Any explanation for why this is? [cid:image001.gif@01CECF2B.581704D0]

2 1

Monitoring events (presence)
by Paul Belanger 22 Oct '13

22 Oct '13

I'm looking for some information about how best to monitor presence events within kamailio? Ideally I'd be writing a python application that will monitor specific endpoints within kamailio and once there state changes I do my thing. I know tapping into the database is an option, but figure I see if there is another method. EG: Opening a websocket to stream events? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 5

Help with NAT
by P. S. 22 Oct '13

22 Oct '13

Hello there, I am trying to setup a Kamailio (3.3) + Linphone + Jitsi based private calling system (users can call each other but not call out of the network). Agents can be behind NAT (think Verizon cellphone or home-users behind their router). I started with the default setup and have tried different settings for NAT/rtpproxy. Currently, all my users register onto the Kamailio server. Users who are not behind a NAT (Jitsi on the internet) i.e. with Public IPs are able to successfully make audio/video calls. I have RTP proxy setup / running as well, and calls to rtpproxy_manage() as required. The process is running and Kamailio is configured to know that the RTPproxy is listening. I had to switch to using TCP on some of the my UAs because of provider blocking UDP traffic. So they are able to register, yet calls are not completed. The call looks like it connects but no media (audio or video goes through). I have read a lot of the forums on NAT - and am trying to solve for a couple of big issues: 1. Figuring out when a UA is behind a NAT, and handling the RTPProxying appropriately. 2. while ngrep-ing UDP packets occassionally I see "Warning: 399 sipalg "Unauthorized" or Bad request. based on what I saw on the internet, it has to do with the service provider (Verizon) messing with the packet. Is there a way arount this? Any help is greatly appreciated! Thanks Pranv

2 1

clearing pua table
by SIP Guru 22 Oct '13

22 Oct '13

Hi, expired entries in pua table won't be deleted. And kamailio keep sending PUBLISH's for expired entries. Am I missing something ? Toni

2 1

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