- sr-users - kamailio.org

Re: [SR-Users] Drouting
by Keith 29 Apr '14

29 Apr '14

Hi all, So I moved away from LCR and dispatcher to use drouting but I can't see how to use probing. In OpenSIPS I can see this is built into the module but it's not in Kamailio. Any ideas how I can get round this? I want to use drouting for all my routing needs. Cheers Keith

2 1

Kamailio and Free switch - multi tenant
by Chirag Desai 29 Apr '14

29 Apr '14

Hi all, First time poster here. I had a freeswitch installation configured to allow users to register with different domains e.g. tenant1.mydomain.com tenant2.mydomain.com .. tenantN.my domain.com This worked fine. Users were able to call extensions in their own domain and access their voicemail. All tenants/domains shared the same diaplan (default). I followed the great tutorial at http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc I now have my users (using different domains as above) registering to Kamailio. It was a little tricky but I can now place calls between users on the same domain. For some reason the variable ${domain_name} (from the tutorial) seems to be empty so I had to replace it with a channel variable (I'm not in front of my dialplan now but it was something like ${sip_from_domain}. Was this the right thing to do? Why does the domain_name variable not populate with the domain of the user making the call? Another issue I have is to do with usernames. I want my users to have alphanumeric usernames e.g. user1(a)tenant1.example.com user2(a)tenant1.example.com But I'd also like them to be able to have an associated extension and mailbox. E.g. user1 may be extension 1001 and mailbox 1001 user2 may be extension 1002 and mailbox 1002. In Kamailio I've handled this by using aliasdb. So user1.tenant1.example.com has an alias of 1001.tenant1.example.com Scenario: User2 dials 1001. Kamailio evaluates this and determines we want user1. It passes the call to FS with a $rU of user1. However, even though user1 is registered and online...the call fails and diverts to voicemail - any ideas? Also I had to hack the above so that after the alias db lookup Kamailio sends an $rU of 1001@tenant1... If anyone has got FS and Kamailio running with multiple domains I would really love some feedback on routing and dialplan configuration. Thanks so much in advance, CD

1 0

kamctl not connect to dbserver
by Юрий Горличенко 28 Apr '14

28 Apr '14

Hello. I started configure Kamailio 3.3 server with mysql database, that running on the remote server. I customise settings at kamailio.cfg (for BDURL) and kamctlc file (for my db - changed user, password, dbhost and other settings, such as dbengine and etc.) Now, when I tried add new subscriber from kamctl util (kamctl add newuser@kamailioIP passwd). I see "access denied for <myuser>@<interface of kamailio server>", But as I aleady says I configured filew with other settings for db server. Where I must configure ip of dbhost to connect kamctl to my dbserver?

2 1

Dispatcher marks TLS destinations as Inactive when present in more than one set
by Giacomo Vacca 28 Apr '14

28 Apr '14

Hi all, I'm seeing a strange behaviour with dispatcher in these conditions: 1. A destination requires TLS (e.g. sips:10.0.0.1:5061) 2. The destination has flags 8 (i.e. in normal conditions we want AP, Active Probing) and dispatcher uses ds_probing_mode 1 (always probe) 3. The destination is present in more than one set The behaviour is that the destination is marked as IP (Inactive Probing) in the set with a lower set number. In other words at startup (or dispatcher reload): SET_NO:: 2 SET:: 2 URI:: sips:10.0.0.1:5061 flags=AP priority=0 attrs= SET:: 1 URI:: sips:10.0.0.1:5061 flags=AP priority=0 attrs= After some time (about 30", see below): SET_NO:: 2 SET:: 2 URI:: sips:10.0.0.1:5061 flags=AP priority=0 attrs= SET:: 1 URI:: sips:10.0.0.1:5061 flags=IP priority=0 attrs= So the same destination is marked as AP and IP at the same time. I've reproduced this with a brand new installation (4.1.3 on debian wheezy), but seen with 3.2.3 and 4.0.4 as well. debian wheezy VM (Linux debian7 3.2.0-4-amd64 #1 SMP Debian 3.2.54-2 x86_64 GNU/Linux) I can see (with either ssldump or asterisk's 'sip set debug on') the first OPTIONS/404 (related to set #2), but not the second (related to set #1). I haven't seen (with tcpdump) the attempt to create a second connection to the destination (which makes me think that it either fails or kamailio attempts to re-use the existing one). Probing for set #1 appears to time out (i.e. after 30" the response is considered a 408). It happens with either Asterisk or FreeSWITCH as servers. Extract from tls.cfg: [client:default] verify_certificate = no require_certificate = no Dispatcher configuration: loadmodule "dispatcher.so" [...] # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.cfg") modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=404") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 30) /etc/kamailio/dispatcher.cfg: 1 sips:10.0.0.1:5061 8 0 2 sips:10.0.0.1:5061 8 0 (I'm using 10.0.0.1 here for simplification only; the real IP address has been omitted). Server (Asterisk or FreeSWITCH) certs are self-signed. Relevant logs (level 4): Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: dispatcher [dispatch.c:2453]: ds_check_timer(): probing set #2, URI sips:10.0.0.1:5061 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sips:10.0.0.1:5061> Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [uac.c:182]: dlg2hash(): DEBUG: dlg2hash: 49688 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: <core> [tcp_main.c:1849]: tcp_send(): tcp_send: no open tcp connection found, opening new one Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 10.0.0.1 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: <core> [tcp_main.c:1096]: tcpconn_new(): tcpconn_new: on port 5061, type 3 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: <core> [tcp_main.c:1408]: tcpconn_add(): tcpconn_add: hashes: 1217:0:0, 1 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSc<default> Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: <core> [tcp_main.c:2737]: tcpconn_1st_send(): pending write on new connection 0x7fd4813cdb70 (-1/225 bytes written) Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [tcp_main.c:3624]: handle_ser_child(): handle_ser_child: read response= 7fd4813cdb70, 5, fd 32 from 9 (11169) Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89dc40, 32, 2, 0x7fd4813cdb70), fd_no=24 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: dispatcher [dispatch.c:2453]: ds_check_timer(): probing set #1, URI sips:10.0.0.1:5061 Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sips:10.0.0.1:5061> Apr 24 15:54:19 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [uac.c:182]: dlg2hash(): DEBUG: dlg2hash: 49689 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [io_wait.h:796]: io_watch_chg(): DBG: io_watch_chg (0x89dc40, 32, 0x1, 0xffffffff) fd_no=25 called Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x89dc40, 32, -1, 0x0) fd_no=25 called Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [tcp_main.c:4302]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 1 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [tcp_main.c:3973]: send2child(): selected tcp worker 0 13(11173) for activity on [tls:192.168.142.170:5061], 0x7fd4813cdb70 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_read.c:1510]: handle_io(): received n=8 con=0x7fd4813cdb70, fd=7 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8e2040, 7, 2, 0x7fd4813cdb70), fd_no=1 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fd4813cdb70 n=198 fd=7 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#001 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_domain.c:712]: sr_ssl_ctx_info_callback(): SSL handshake done Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL disable renegotiation Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_server.c:413]: tls_connect(): TLS connect successful Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_server.c:420]: tls_connect(): tls_connect: new connection to 10.0.0.1:5061 using TLSv1/SSLv3 AES256-SHA 256 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_server.c:423]: tls_connect(): tls_connect: sending socket: 192.168.142.170:43914 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_server.c:305]: tls_dump_cert_info(): tls_connect: server certificate subject:/CN=REMOVED/O=REMOVED Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_server.c:309]: tls_dump_cert_info(): tls_connect: server certificate issuer:/CN=REMOVED/O=REMOVED Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_server.c:429]: tls_connect(): WARNING: tls_connect: server certificate verification failed!!! Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tls [tls_dump_vf.c:105]: tls_dump_verification_failure(): verification failure: self signed certificate in certificate chain Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fd4813cdb70 n=805 fd=7 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=[...REMOVED...] Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_read.c:1317]: tcp_read_req(): tcp_read_req: content-length= 0 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <404> Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Not Found> Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK812c.b064d3e3000000000000000000000000.0>; state=6 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 234, <received> = <REMOVED>; state=6 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <59525>; state=16 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=as232959d2 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [43]; uri=[sips:10.0.0.1:5061] Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sips:10.0.0.1:5061>] Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <OPTIONS> Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 49688 label 0 branch 0 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0x7fd4813cc530)! Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=1 T end=0x7fd4813cc530 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_reply.c:2206]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=2 is_invite=0) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=404 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7fd4813cc530, callback type 1024, id 0 entered Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: dispatcher [dispatch.c:2392]: ds_options_callback(): OPTIONS-Request was finished with code 404 (to sips:10.0.0.1:5061, group 2) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: tm [t_reply.c:1663]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Apr 24 15:54:20 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [receive.c:296]: receive_msg(): receive_msg: cleaning up Apr 24 15:54:26 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e2040, 7, -1, 0x10) fd_no=2 called Apr 24 15:54:26 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7fd4813cdb70, state 1, fd=7, id=1 Apr 24 15:54:26 debian7 /usr/sbin/kamailio[11173]: DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7fd4813b51f0 Apr 24 15:54:26 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7fd4813cdb70, 1 from 0 Apr 24 15:54:26 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89dc40, 32, 2, 0x7fd4813cdb70), fd_no=24 Apr 24 15:54:26 debian7 /usr/sbin/kamailio[11177]: DEBUG: <core> [tcp_main.c:3515]: handle_tcp_child(): handle_tcp_child: CONN_RELEASE 0x7fd4813cdb70 refcnt= 1 Apr 24 15:54:49 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Apr 24 15:54:49 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 Apr 24 15:54:49 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed Apr 24 15:54:49 debian7 /usr/sbin/kamailio[11169]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7fd4813e5890, callback type 1024, id 0 entered Apr 24 15:54:49 debian7 /usr/sbin/kamailio[11169]: DEBUG: dispatcher [dispatch.c:2392]: ds_options_callback(): OPTIONS-Request was finished with code 408 (to sips:10.0.0.1:5061, group 1) Probing set #2 goes OK. Probing set #1: starts probing at 15:54:19, and times out 30" later at 15:54:49. Is this a bug, a misconfiguration, or perhaps a wrong way to manage dispatcher sets using TLS? Additional question: Is it not expected that a persistent TLS connection is established with a 'sips' dispatcher destination? Thanks in advance for the time to read through all this! Giacomo Truphone Limited, registered in England and Wales (registered company number: 04187081). Registered office: CityPoint, One Ropemaker Street, London EC2Y 9SS. VAT No. GB 851 5278 19 This e-mail, and any attachment(s), may contain information which is confidential and/or privileged, and is intended for the addressee only. If you are not the intended recipient, you may not use, disclose, copy or distribute this information in any manner whatsoever. If you have received this e-mail in error, please contact the sender immediately and delete it.

2 2

Early dialog
by Ivan Baques Garcia 28 Apr '14

28 Apr '14

Hi, I'm using modparam("dialog", "db_mode", 1), but only confirmed dialogs are shown in the database. How I can enable to show unconfirmed dialogs in the database? Now are only in-memory. Best regards

3 2

Large Content-Length crash with instant message
by Roberto Fichera 28 Apr '14

28 Apr '14

Hi All, I'm playing with instant messaging with a PJSIP client, but as soon as I send body with payload to about 100k I'm getting the error below: Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: <core> [tcp_main.c:715]: _wbufq_add(): ERROR: wbufq_add(40552 bytes): write queue full or timeout (0, total 0, last write 51946505 s ago) Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: tm [../../forward.h:240]: msg_send(): msg_send: ERROR: tcp_send failed Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2910]: ERROR: <core> [tcp_main.c:3638]: handle_ser_child(): handle_ser_child: ERROR: received CON_ERROR for 0xb30ddcd8 (id 352), refcnt 2, flags 0x5092 Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 28 09:16:52 ip-10-227-0-26 /usr/sbin/kamailio[2909]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) And the app get SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/SL) Via: SIP/2.0/TLS XX.XX.XX.XX:55859;rport=55859;branch=z9hG4bKPjkkazEy.XcjJkYDCEzwbwUVAONDzcCN8t;alias From: <sip:aaaaa@test.net>;tag=9gcb.Ay3wSCZNFgs8PNCbdkRq7jfxbDx To: <sip:bbbbb@test.net>;tag=9a285338e10af1c086406d08dc7f5d79.e457 Call-ID: v9JlsK1AEH4So8AIEfqiGuUOEx5Fub1C CSeq: 28450 MESSAGE Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 --end msg-- Kamailio is already set with tcp_rd_buf_size=204800 Does anyone can suggest how to fix this? Thanks in advance, Roberto Fichera.

2 4

LCR from_uri help
by Juha Heinanen 28 Apr '14

28 Apr '14

just to complete this thread, it turned out that from uri matching didn't happen because load_gws() was called without caller uri argument. -- juha

1 0

Presence request authentication
by Muhammad Shahzad 28 Apr '14

28 Apr '14

Hi, I have a complex setup consisting of two sip server, lets call them main server and presence server. The main server manages SIP register, calls, messages and so, however it does not support presence at all. It returns SIP response "405 Method Not Allowed" for any SIP PUBLISH, SUBSCRIBE or NOTIFY. The presence server is a Kamailio server, which unfortunately has no access to SIP subscriber database (so it can not do authentication), however it has access to User Location database. Now at presence server i want to put some kind of security for presence requests, since i can't do authentication, so i was thinking if i can verify if request has come from a location registered in location db. I found method "registered" in registrar module but unfortunately it checks if R-URI or any given URI is registered, while i want to do check against "received" address or contact address. Can you guys suggest any easy way to do it? I am thinking about using "reg_fetch_contacts" but not sure if it can serve the purpose. Thank you.

2 2

Kamailio and FHoSS integration
by Andrea Meroni 28 Apr '14

28 Apr '14

Hi all, i am testing a very basic IMS core Kamailio-based using as HSS FHoSS. However, while using SIPp to stress the system, i noticed that the VM hosting the FHoSS is overloaded (even with 2 registration scenarios per second), in particular using top i was able to see that the java process used almost 200% of the CPU (i run the VM using qemu with options -m 1024 -smp 4). Is it normal or have i misconfigured something? Someone else has the same problem? Andrea

4 7

$(hdr(name)[N]) question
by Juha Heinanen 28 Apr '14

28 Apr '14

wiki was this related to $(hdr(name)[N]): If [N] is omitted then the body of the first header is printed. The first header is got when N=0, for the second N=1, a.s.o. In case of a comma-separated multi-body headers, it returns all the bodies, comma-separated. i don't understand the last sentence. how can i get all bodies of all headers of name=x as a comma separated list? -- juha

2 8

jsSIP client disconnects after a while when using Kamailio websocket
by Alex Villacís Lasso 28 Apr '14

28 Apr '14

I am having this issue when using kamailio-4.1.2 (after patching the segfault bug). The kamailio is configurated as a frontend for asterisk. When using UDP for SIP, this works correctly. However, I am now setting up websockets (no TLS yet), and I am getting a disconnect after an interval. I am getting this on the logs: Apr 11 16:19:41 elx3 /usr/sbin/kamailio[4814]: WARNING: websocket [ws_frame.c:868]: ws_keepalive(): forcibly closing connection Tweaking with keepalive_mechanism=1 and keepalive_timeout does not change the situation, and does not even modify the interval before the disconnection. Is this keepalive something that should be visible on a SIP trace? I am only seeing REGISTER and its response from the jsSIP trace. I do not want to have to disable keepalives just to work around this issue.

2 1

480 Temporarily Unavailable
by David Giannetto 28 Apr '14

28 Apr '14

I get this error "480 Temporarily Unavailable" Every time I dial 206-456-1041 this is the information I have entered in the the filds: line 1 Username: optimized72 <optimized72(a)iptel.org> Password: Freek101@ Proxy: iptel.org -- *l <http://LEDSLEDSLEDS.COM>edslesdlesd.com <http://edslesdlesd.com>* David Giannetto VOIP COM 915 235 47 86 105 E San Antonio Ave Suit 303 El Paso, Texas, 79901 VOIP COM 625 106 6388 Lerdo y 19 # 986 CD. Cuauhtemoc, Chih.

2 1

Translate calling/called party based on username to telephone number base id.
by Helena Garcia-Nieto 28 Apr '14

28 Apr '14

Hi all. I want to thank you all for the great info and help that you provide in this mailing list. I am just starting to work with kamailio and still learning. I would like some help to point me on the right direction on our new challenge: We have a kamailio server acting basically as location server and call proxy. We are connected to a PSTN gw to reach the outside world. It is working fine but now the client requested some changes that may imply we work with username base subscribers (user1. Characters based) at kamailio side but we still need to convert these usernames to numbers when sending the call to the PSTN gw. Username-number should be fix and can be stored in a table on the db. We would need to lookup userpart of all the header (from, to, contact.) and substitute incoming username with the peer number for calls from the app to pstn. And translate numbers to usernames on the calls from pstn to the apps. I am reading uac module and the sqlops module to search in the db before the substitution but I am not sure if that is the best approach to the problem. Could anyone point me in the right direction or suggest a better way to do that? Have anyone already implemented this type of modifications? I am a bit worried for the dialog consistency although I've read the uac module should take care of it Thanks in advanced Helena From: Helena Garcia-Nieto [mailto:helena.gnieto@morodo.co.uk] Sent: martes, 10 de diciembre de 2013 16:19 To: 'Kamailio (SER) - Users Mailing List' Cc: helena.gnieto(a)morodo.co.uk Subject: 500 I'm terribly sorry error Hello, Thanks in advanced for the help. I am almost new with kamailio and still struggling through silly problems so please forgive me if the solution is so obvious. I have a network like Xlitle -- Kamailio -- GW The GW is more or less out of my reach for changing the behaivour. As devices I have xlitle Kamailio is on version 4.0.2 I've changed only few things from the default config file. Add mysql support, auth, userlocdb, pstngw. For this part, gw routing , I've defined gw ip and port inside the PSTN definition like: #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" iskratel.gw_ip = "10.XX.XX.XX" iskratel.gw_port = "5060" #!endif I route the calls with: route(ISKRATEL); And defined a routing function route[ISKRATEL] { #!ifdef WITH_PSTN # check if ISKRATEL GW IP is defined if (strempty($sel(cfg_get.iskratel.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but iskratel.gw_ip not defined\n"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.iskratel.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.iskratel.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.iskratel.gw_ip) + ":" + $sel(cfg_get.iskratel.gw_port); } # Add profix to ISKRATEL: A99901 subst_uri('/^sip:(.*)/sip:A99901\1/i'); # add A99901 route(RELAY); exit; #!endif return; } I can see that the code goes all the way to the route(relay) but all I can see is the 500 I'm terribly sorry, server error occurred (7/SL) and a second response 500 I'm terribly sorry, server error occurred (7/TM) I've tried defining the IP as the given pstn.gw_ip and route(pstn) without changing anything on the pst default routing but the response of the server is the same. I cannot see any special error on the logs. I have the exact same config for this part, in another test server and the calls go to the gw without reporting any error. I'll appreciate any help from you! Thanks in advanced for your time Helena

2 1

test if record_route() has been called already?
by Juha Heinanen 28 Apr '14

28 Apr '14

is there any means to test if record_route() has been called already on the request? if it is called more than once, error message is issued. -- juha

2 1

Processing on Invite Packet
by aawaise 28 Apr '14

28 Apr '14

Hello, On arrival of INVITE packet to the Kamailio server, which module is used to take in response to Invite packet. Other than record_route, which records the route. And what to do if we want to add some actions in response to INVITE packet. Cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Processing-on-Invite-Packet-tp12709… Sent from the Users mailing list archive at Nabble.com.

2 1

DRouting, routeid is not triggered
by Maciej Bylica 28 Apr '14

28 Apr '14

Hello, I am working on version: kamailio 4.1.2 (x86_64/linux) and heaving troubles with drouting module. The problem i am facing is that kamailio cannot enter routeid that is provided inside dr_rule table. My last try was to use standard script file with placing following data before t_relay if (!do_routing("10")) { sl_send_reply("403", "No route for You"); exit; } additionaly i have added route definition route[1] { xlog("L_INFO","[INFO] Default route --- fn-$fn, fu-$fu, fU-$fU, ru-$ru, rU-$rU, sp-$sp, si-$si, tu-$tu, tU-$tU"); uac_replace_from("sip:$fU@10.10.10.5"); } My DB contains following data: +--------+---------+--------+---------+----------+---------+--------+--------------+ | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | description | +--------+---------+--------+---------+----------+---------+--------+--------------+ | 1 | 10 | 43 | | 0 | 1 | 1 | test rule | +--------+---------+--------+---------+----------+---------+--------+--------------+ +------+------+---------------+-------+------------+-------+-------------+ | gwid | type | address | strip | pri_prefix | attrs | description | +------+------+---------------+-------+------------+-------+-------------+ | 1 | 10 | 10.10.10.9 | 0 | NULL | NULL | FirstGW | +------+------+---------------+-------+------------+-------+-------------+ The result is that script is omitting route[1] without any purpose (xlog is not shown, $fU is not modified). Could somebody help me to understand whats going on? Thanks in advanced, Mac

2 8

avpops question...
by arun Jayaprakash 28 Apr '14

28 Apr '14

Hello, I have a piece of code as follows: $avp(callfwdcount) = 0; $avp(dnumber) = $rU; $var(found) = 0; while($avp(callfwdcount) < 3) { if (avp_db_load("$avp(dnumber)", "$avp(callfwd_permanent)")) { ..... $avp(dst_number) = $avp(callfwd_permanent); # SET THE CALL FORWARD TO NUMBER .... } Now this works if this is a single domain set up. I am running a multidomain set up and need to check the domain part to ( $rd ) and if there is match in the database with $rU and $rd then the forward to number needs to be set. Can someone let me know how this can be done? Thank you, Arun

2 2

Dialog module and confirmed-unacked dialogs
by Alex Balashov 28 Apr '14

28 Apr '14

Hi, I am wondering if perhaps we ought to do something with regard to specific handling of confirmed-nonacked dialogs (CONFIRMED_NA state in dlg_hash.h) in the dialog module. These are dialogs where a 2xx reply is sent to the opening INVITE transaction, but no end-to-end ACK is seen[1] by Kamailio and thus the dialog is not recorded as transitioning to CONFIRMED state. This can happen for a variety of reasons, but the most common scenario I run into is the CANCEL-200 OK race, where the caller cancels the call just as the callee answers it, near-simultaneously. The 200 OK hasn't gotten back to the caller yet, so when it receives it, it has no effect, because from the caller's point of view, the dialog has already been CANCEL'd. Meanwhile, the CANCEL has no effect on the callee end either, since, from its point of view, the dialog has already transitioned into confirmed state. The problem I am running into a lot is that these dialogs stay tracked up until the dialog timeout period, which can be several hours away. In high-volume environments, they can clog up concurrent channel counts. The receiving UAS has, of course, disposed of these dialogs long ago, after 64*T1, but they remain "stuck" in Kamailio. I know that RFC 3261 Section 13.3.1.4 ("The INVITE is Accepted") says: If the server retransmits the 2xx response for 64*T1 seconds without receiving an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is accomplished with a BYE, as described in Section 15. Now, I know that "SHOULD" != "MUST", and I would imagine that this is probably the main reason why the dialog module does not time out such dialogs according to the same timers as the callee UA might. Nevertheless, they present a problem. Right now, I deal with it by using a script that combs 'kamctl fifo dlg_list' for dialogs in 'state:: 3' for more than X seconds and manually ends them. But, when there's a problem that one runs into nearly ubiquitously in all deployments with nontrivial deployments, it seems to me it's time to consider an additional 'dialog' modparam or something of that ilk that can provide an expedited timeout for nonacked dialogs. I would be happy to write such a patch. The reason I am bringing it up to the community is because I am uncertain as to whether this might have any unforeseen consequences, or whether it's been discussed in various dialog_ng discussions in the past that I have not carefully monitored. Thanks! -- Alex (With apologies for cross-posting.) [1] Or correctly associated based on tight matching. -- 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

permanent 1.5 s timer interval in module pipelimit
by Julia 28 Apr '14

28 Apr '14

Hello all, Timer pl_timer in pipelimit.c is defined as MS_TO_TICKS(1500) instead of timer_interval (in versions 3.3 - 4.1). I found a patch that solves the same problem in the ratelimit module: Module: sip-router Branch: 3.1 Commit: f027ca95a2a1818cd40e4ed6d4358a76d18252ad URL: <http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=f027ca9 5a2a1818cd40e4ed6d4358a76d18252ad> http://git.sip-router.org/cgi-bin/gitweb.cgi/... Author: Marius Zbihlei Committer: Marius Zbihlei Date: Mon Apr 11 13:45:10 2011 +0300 modules/ratelimit Timer was ran at a 1.5 s interval instead of timer_interval, thus rendering TAILDROP alg unusable (cherry picked from commit 3a06d7d3561f889c84249eaa5ec7a5b79a2a23bd) Regards, Julia

2 1

kamctl not connect to dbserver
by Юрий Горличенко 28 Apr '14

28 Apr '14

Hello. I started configure Kamailio 3.3 server with mysql database, that running on the remote server. I customise settings at kamailio.cfg (for BDURL) and kamctlc file (for my db - changed user, password, dbhost and other settings, such as dbengine and etc.) Now, when I tried add new subscriber from kamctl util (kamctl add newuser@kamailioIP passwd). I see "access denied for <myuser>@<interface of kamailio server>", But as I aleady says I configured filew with other settings for db server. Where I must configure ip of dbhost to connect kamctl to my dbserver?

2 1

kamailio command line
by betul.cerit＠netgsm.com.tr 28 Apr '14

28 Apr '14

Hi to all, I’m using “kamailio” command to see alerts,notices and warnings but it also starts a new kamailio process.Is there a way to see command line without starting a new kamailio process? I need something just like asterisk –rvvvvvvvvvvv in asterisk. Thanks in Advance.

2 2

No secure attributes from rtpengine in SRTP/RTP bridge mode
by Alexey Rybalko 28 Apr '14

28 Apr '14

Hello! I have been experimenting with drop-in replacement of old "rtpproxy-ng" module with new "rtpengine". Wondering what is wrong in my configuration: there are no security attributes in rtpengine answer on RTP/SAVPF offer. Neither "fingerprint" (DTLS) nor "crypto" (SDES). I used Firefox 29 during this test. 1. Here's original offer : INVITE sip:user5@..... SIP/2.0 > Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 > Max-Forwards: 69 > To: <sip:user5@......> > From: "user4" <sip:user4@......>;tag=dvf1co8urv > Call-ID: phmq9o62cv21timhfnpf > CSeq: 4233 INVITE > Proxy-Authorization: Digest algorithm=MD5, username="user4", > realm="......", nonce="U1o8H1NaOvP3wxegsYCKOJX7S7DV/r1N", uri="sip:user5@......", > response="44e7b16c55d3237f63e04b3c0b194f45" > Contact: <sip:user4@ > ......;gr=urn:uuid:c193bcd4-aa2e-47ef-a106-22e60f5fde9e;ob> > Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE > Content-Type: application/sdp > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.7 > Content-Length: 607 > > v=0 > o=Mozilla-SIPUA-29.0 15825 0 IN IP4 0.0.0.0 > s=SIP Call > t=0 0 > a=ice-ufrag:2102f082 > a=ice-pwd:8733e5248a7fb087b40ea45b3ca6f634 > *a=fingerprint:sha-256 > 32:AA:85:DB:D8:3C:E6:C3:46:07:11:9E:9F:54:B9:42:7F:5C:37:5F:9D:D1:AD:19:22:A3:AC:9C:6C:A5:A6:CD* > m=audio 62290 *RTP/SAVPF* 109 0 8 101 > c=IN IP4 ..... > a=rtpmap:109 opus/48000/2 > a=ptime:20 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > ..... > Here's the snippet for translation SRTP-RTP: rtpengine_offer("force trust-address symmetric replace-origin replace-session-connection ICE=force_relay *RTP/AVP*"); 2. Here's final answer (from rtpengine): SIP/2.0 200 OK > Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 > Record-Route: <sip:......;lr;nat=yes> > Record-Route: <sip:.....:15060;transport=udp;lr;ovid=3207d8cd> > Record-Route: <sip:95f6551e81@.......:10080;transport=ws;lr;ovid=3207d8cd> > Contact: <sip:user5@......1:49362> > To: <sip:user5@.....>;tag=4d436110 > From: "user4"<sip:user4@....>;tag=dvf1co8urv > Call-ID: phmq9o62cv21timhfnpf > CSeq: 4233 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces, eventlist > > User-Agent: X-Lite release 4.5.4 stamp 70864 > Content-Length: 432 > > v=0 > o=- 1398422264455879 3 IN IP4 ..... > s=X-Lite 4 release 4.5.4 stamp 70864 > c=IN IP4 ...... > t=0 0 > m=audio 30002 *RTP/SAVPF* 109 0 8 101 > a=rtpmap:109 opus/48000/2 > a=fmtp:109 useinbandfec=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=rtcp:30003 > .... > Error from JsSIP: {name: "INTERNAL_ERROR", message: "Could not negotiate answer SDP; cause = > *NO_DTLS_FINGERPRINT*", __exposedProps__: Object} > Here's the snippet for translation RTP-SRTP: rtpengine_answer("force trust-address symmetric replace-origin replace-session-connection rtcp-mux-demux ICE=force *RTP/SAVPF* "); There was another test with Chrome 34 with the same result. Offer: INVITE sip:user5@.... SIP/2.0 Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 Max-Forwards: 69 To: <sip:user5@....> From: "user4" <sip:user4@....>;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: 957 INVITE Proxy-Authorization: Digest algorithm=MD5, username="user4", realm=".....", nonce="U1pKZ1NaSTtLo2vjK9TGyBu6Axb+EtyN", uri="sip:user5@...", response="98f6275d3636664b611ff1411af982af" Contact: <sip:user4@....;gr=urn:uuid:8cd4e797-7314-485f-b191-4a15a6581c42;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 1586 v=0 o=- 7510391807340598328 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS ErnvtgCDe9aR6LWxNRgT83r0mxtyAV87LUxT m=audio 51672 *RTP/SAVPF *111 103 104 0 8 106 105 13 126 c=IN IP4 10.61.2.151 a=rtcp:51672 IN IP4 .... ...... a=ice-ufrag:EMn7uHfSS7ulRGU2 a=ice-pwd:FskTdhj7qT6ELP7uTIb+gquQ a=ice-options:google-ice *a=fingerprint:sha-256 46:6E:E0:18:4A:C5:06:A8:26:85:ED:FE:16:C1:86:5E:8D:BC:4D:D9:F2:1A:75:81:A1:A7:CE:5A:79:4D:B7:22*a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux *a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:R6qaiU7Cm471zNF6f3Q87TyXbHjEt/VhLgUgY2ZZ a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lch+mfN/hi9QmseWu+ss1M2vA8mwRh8GQYChaJvc* a=rtpmap:111 opus/48000/2 .... Answer: SIP/2.0 200 OK Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 > Record-Route: <sip:...;lr;nat=yes> Record-Route: > <sip:....:15060;transport=udp;lr;ovid=3207d8cd> Record-Route: > <sip:712a450958@...:10080;transport=ws;lr;ovid=3207d8cd> Contact: < > sip:user5@10.61.2.151:49362> To: <sip:user5@....>;tag=b8c8ee57 From: > "user4"<sip:user4@...>;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: > 957 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, > MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, > eventlist User-Agent: X-Lite release 4.5.4 stamp 70864 Content-Length: 432 > v=0 o=- 1398425917116411 3 IN IP4 ... s=X-Lite 4 release 4.5.4 stamp 70864 > c=IN IP4 .... t=0 0 m=audio 30006 *RTP/SAVPF* 111 0 8 126 a=rtpmap:111 > opus/48000/2 a=fmtp:111 useinbandfec=1 a=rtpmap:126 telephone-event/8000 > a=fmtp:126 0-15 a=sendrecv a=rtcp:30007 .... > Error is JsSIP: Failed to set remote answer sdp: Called with a SDP without crypto enabled > Any help on this issue would be greatly appreciated! best regards, Alexey

3 7

Double attempt to record-route
by Juha Heinanen 27 Apr '14

27 Apr '14

I tried to call record_route_preset() in branch route in order to add contact specific RR headers when request is forked to more than one contact. What happened was that I got error message: Sep 26 17:03:47 wheezy2 /usr/sbin/sip-proxy[2436]: ERROR: rr [rr_mod.c:256]: Duble attempt to record-route Is it really so that record routing is not a branch specific thing? Is there a way to solve this problem? -- Juha

3 4

Kamailio+IMS+WebSocket
by mohammed alyaseen 26 Apr '14

26 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo Mohammed Al-khameri Technical University Berlin TUB

1 0

Re: [SR-Users] cnxcc
by Andrzej Ciupek 26 Apr '14

26 Apr '14

Hello I’am using default config with options from an example. It looks like event route isn’t execuded after hangup. Apr 26 16:26:14 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:15 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:16 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:17 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:18 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:19 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: Setting up money based credit control Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: Setting up channel based credit control Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: CNXCC ROUTE: 1000 has 2 call(s) kamailio.cfg … … # ----- prepaid call controll ----- #!define DLG_FLAG 28 #!define CC_FLAG 29 loadmodule "dialog.so" modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") loadmodule "cnxcc.so" modparam("cnxcc", "dlg_flag", CC_FLAG) modparam("cnxcc", "credit_check_period", 1) #check every 1 second … … … route { … … if (!allow_source_address("2") and !allow_source_address("1")){ route(CNXCC); route(PSTN); } … } route[CNXCC] { $var(client) = "1000"; xlog("L_INFO", "Setting up money based credit control"); $var(credit) = "10"; # 10$ of credit $var(cost_per_sec) = "1"; # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second # if only one call is established, that call should last 9 seconds. if (!cnxcc_set_max_credit("$var(client)", "$var(credit)", "$var(cost_per_sec)", "$var(i_pulse)", "$var(f_pulse)")) { xlog("Error setting up credit control"); return; } $var(count) = -1; if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { xlog("Error getting customer's channel count"); } xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)"); if ($var(retcode) < -1) { xlog("Too many channels for customer"); sl_send_reply(403, "Forbidden"); if (!cnxcc_terminate_all("$var(client)")) { xlog("Error terminating customer's calls"); } else { xlog("Terminating customer's call"); } exit; } } route[SHOW_CHANNEL_COUNT] { $var(count) = @cnxcc.channels["1000"].count; xlog("L_INFO", "1000 has $var(count) call(s)"); } event_route[dialog:failed] { xlog("dialog failed"); } event_route[cnxcc:call-shutdown] { xlog("L_INFO", "Shutting down channel by callcontroll"); xlog("L_INFO", "[$ci]: call killed"); $var(count) = $var(count)-1; if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { xlog("Error getting customer's channel count"); } xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)"); # perform some kind of notification, database update, email sending, etc } Greetings Andrzej Ciupek Od: Carlos Ruiz Díaz Wysłano: ‎sobota‎, ‎26‎ ‎kwietnia‎ ‎2014 ‎15‎:‎19 Do: Kamailio (SER) - Users Mailing List, Andrzej Ciupek Hi, sorry guys, I forgot about this. I will make my own tests and post the results. Andrzej, can you post the configuration file you are using? Regards, Carlos On Sat, Apr 26, 2014 at 4:15 AM, Kelvin Chua <kelchy(a)gmail.com> wrote: No, I have not yet found the solution for this issue on my setup. Maybe Carlos can help us out On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> wrote: Hello Have You solved this issue ? I’am using Kamailio 4.1.3 and I’am getting the same results. Greetings Andrzej Ciupek _______________________________________________ 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 -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

2 1

Re: [SR-Users] cnxcc
by Kelvin Chua 26 Apr '14

26 Apr '14

No, I have not yet found the solution for this issue on my setup. Maybe Carlos can help us out On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> wrote: > Hello > > Have You solved this issue ? > I’am using Kamailio 4.1.3 and I’am getting the same results. > > Greetings > Andrzej Ciupek >

2 1

Kamailio+IMS+WebSocket
by mohammed alyaseen 26 Apr '14

26 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo Mohammed Al-khameri Technical University Berlin TUB

1 0

LCR from_uri help
by Geoffrey Mina 26 Apr '14

26 Apr '14

Greetings, I need a little help with the LCR module. I am running Kamailio 4.1.2. The LCR module is behaving in a way that is inconsistent with our old 1.5 versions. We have a customer that passes a certain caller-id. That from_uri should be routing over a specific trunk. Here are the relevant bits of the LCR data from kamcmd. I would expect gw_index: 7 to match before gw_index: 6 here, but that is not the case. Any help would be greatly appreciated. Please note, i Have stripped out a bunch of other routes that are not in play here... hence the reason gw_index starts at "6". kamcmd> lcr.dump_gws { lcr_id: 1 gw_id: 11 gw_index: 6 gw_name: Route A scheme: sip ip_addr: 1.2.3.4 hostname: port: 5060 params: transport: UDP strip: 0 prefix: tag: flags: 2 defunct_until: 0 } { lcr_id: 1 gw_id: 1 gw_index: 7 gw_name: Route P scheme: sip ip_addr: 4.3.2.1 hostname: port: 5060 params: transport: UDP strip: 0 prefix: DLR# tag: flags: 0 defunct_until: 0 } kamcmd> lcr.dump_rules { lcr_id: 1 rule_id: 31 prefix: +1 from_uri: 2089328663 request_uri: stopper: 0 } { gw_index: 6 priority: 1 weight: 1 } { lcr_id: 1 rule_id: 5 prefix: +1 from_uri: request_uri: stopper: 0 } { gw_index: 7 priority: 1 weight: 1 }

2 7

Kam 4.1 Crash: HELP PLZ!!!!
by Samuel Ware 26 Apr '14

26 Apr '14

Here are the logs from the first crash Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7feac6836310,65572,0,205.251.172.14:5060,16): Message too long(90) Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:14:05 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7feaace02c98,65533,0,205.251.172.14:5060,16): Message too long(90) Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:14:20 tel-vc-fs03 /usr/local/sbin/kamailio[7594]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7feac6836310,65569,0,205.251.172.14:5060,16): Message too long(90) Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:14:27 tel-vc-fs03 /usr/local/sbin/kamailio[7583]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 13:14:32 tel-vc-fs03 /usr/local/sbin/kamailio[7589]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7feaca6c9738), called from <core>: mem/shm_mem.c: sh_realloc(88), first free <core>: mem/shm_mem.c: sh_realloc(88) - aborting Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: ALERT: <core> [main.c:775]: handle_sigs(): child process 7589 exited by a signal 6 Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7604]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7597]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7579]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7603]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7601]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Logs from second crash Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f34ec4f6ed8,65569,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 14:38:42 tel-vc-fs03 /usr/local/sbin/kamailio[3979]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f35242e5d28,65535,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 14:38:44 tel-vc-fs03 /usr/local/sbin/kamailio[3965]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f3516fb6720,65578,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f37d8e91088,65522,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: sl [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f37d8e91088,65522,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:57 tel-vc-fs03 /usr/local/sbin/kamailio[3968]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3961]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f37d81fa960,65522,0,205.251.172.14:5060,16): Message too long(90) Apr 3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3961]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3974]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f34d7d6b720), called from <core>: mem/shm_mem.c: sh_realloc(88), first free <core>: mem/shm_mem.c: sh_realloc(88) - aborting Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3935]: ALERT: <core> [main.c:775]: handle_sigs(): child process 3974 exited by a signal 6 Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3935]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3935]: INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3985]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3983]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3966]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3988]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3972]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received Apr 3 14:39:20 tel-vc-fs03 /usr/local/sbin/kamailio[3984]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received

5 10

Kamailio+IMS+WebSocket
by mohammed alyaseen 25 Apr '14

25 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo

1 0

Re: [SR-Users] Minimalist Kamilio configuration example using dynamic routing module
by Gary Wallis 25 Apr '14

25 Apr '14

Hi, We are moving away from OpenSIPS and would like to start testing Kamailio. I really liked how Kamailio complies so clean and is configured easily in comparison with OpenSIPS monstrosity. So to not waste time can anybody provide some practical info for... Kamailio complilation and config examples (w/MySQL) to route Carrier SIP (DID incoming and outgoing PSTN termination) traffic to and from Asterisk PBXs. Example current OpenSIPS setup using dynamic routing module: 1-. dr_rules have the complete DID for PBXs (for incoming traffic from Carriers to be proxied to the correct PBX). 2-. dr_rules have the partial DID for Carrier gateways (for example based on internatinal, state, etc. routing of outgoing traffic from PBXs to Carriers). 3-. dr_gateways have the IP numbers for PBXs. 4-. dr_gateways have the IP numbers for Carriers. Any info and pointers appreciated. Best regards, Gary

2 2

Re: [SR-Users] Caller ID fields incorrect on Polycom 650s
by J Lee 25 Apr '14

25 Apr '14

Thanks for your response Daniel! I think I omitted a key piece of information in my original question! We are using the SCA module. Hope that sheds some more light on the situation. -jl > Date: Thu, 24 Apr 2014 23:06:28 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Caller ID fields incorrect on Polycom 650s > Message-ID: <53597CD4.8030106(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > kamailio is not touching from/to headers by default. If there are > incorrect, then could be somewhere else. You ahve tos end a sip trace > (ngrep for example) with incoming and outgoing sip packets on kamailio > server. > > Cheers, > Daniel > > On 24/04/14 21:58, J Lee wrote: >> Hi there. >> >> I have a situation where the caller id information on my phone appear >> incorrectly. >> When I have ext 1 dial ext 2, on the caller's phone, instead of saying >> "To: 2", it says "From: 2". >> >> I'm using kamailio version 4.0.4. I'm just new to everything kamailio >> and VoIP so I apologize in advance for any remedial questions. >> >> So far, I've done a SIP trace to and checked all packets line by line >> to see if I could spot where the To address was erroneously changed to >> FROM. In particular, I was checking the call-info fields for caller >> id data or any other data what would appear on the LED of the phone. >> >> Now I'm wondering, who's job (aka which module) is it to set the >> caller id information for both phones? >> I'm thinking of digging inside the code for the right module just to >> see if I can follow what's going on >> >> In addition to answering the question above, if you have any >> suggestions on what else I can check or try, I'd appreciate it. >> >> Thanks List! >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v4.1.3 Released
by Daniel-Constantin Mierla 25 Apr '14

25 Apr '14

Hello, Kamailio SIP Server v4.1.3 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.2. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.3. For more details about version 4.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/04/kamailio-v4-1-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Caller ID fields incorrect on Polycom 650s
by J Lee 24 Apr '14

24 Apr '14

Hi there. I have a situation where the caller id information on my phone appear incorrectly. When I have ext 1 dial ext 2, on the caller's phone, instead of saying "To: 2", it says "From: 2". I'm using kamailio version 4.0.4. I'm just new to everything kamailio and VoIP so I apologize in advance for any remedial questions. So far, I've done a SIP trace to and checked all packets line by line to see if I could spot where the To address was erroneously changed to FROM. In particular, I was checking the call-info fields for caller id data or any other data what would appear on the LED of the phone. Now I'm wondering, who's job (aka which module) is it to set the caller id information for both phones? I'm thinking of digging inside the code for the right module just to see if I can follow what's going on In addition to answering the question above, if you have any suggestions on what else I can check or try, I'd appreciate it. Thanks List!

2 1

Problem accessing git repository
by arun Jayaprakash 24 Apr '14

24 Apr '14

Is there a problem with the git repository? When I try to download kamailio (git clone --depth 1 git://git.sip-router.org/sip-router kamailio ) I get an error saying that git.sip-router.org can not be reached. Please let me know, thank you. Arun

2 2

Re: [SR-Users] TCP reset
by Ciprus, Daniel 24 Apr '14

24 Apr '14

Understand :-) thank you Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Carsten Bock Date:04/24/2014 6:09 AM (GMT-05:00) To: "Ciprus, Daniel" Cc: "Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] TCP reset Hi Daniel, Define "near future"... :-) Once we have time, we may look at this topic; but no timeline yet.... Kind regards, Carsten 2014-04-23 15:55 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>: > Thanks Carsten, > > Any plans to do so in near future ? :-) > > On 04/23/2014 09:52 AM, Carsten Bock wrote: > > Hi Daniel, > > what you want, is basically already implemented in the "Standard" > usrloc module, see: > > http://kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.handle_… > > We haven't implemented that feature yet in the IMS modules.... > > Schöne Grüße, > Carsten > > > 2014-04-23 15:47 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>: > > Hi, > > Is there any way to remove registration entry from registrar (Kamailio IMS > built on stable 4.1) detected on p-cscf by incoming TCP RST from the network > ? What's happening is that client is loosing connections/crashing and TCP > RST is sent back to pcscf which is not propagating changes back to > registrar. This is likely not possible with my version of kamailio but at > least logging which would indicate change of the status will be helpful. > > thanks for any hints > > > -- > Daniel Ciprus > Integration engineer > http://www.acision.com > > 9954 Mayland Dr > Suite 3100 > Richmond, VA 23233 > USA > T: +1 804 762 5601 > E: daniel.ciprus(a)acision.com > > ________________________________ > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you for > understanding. > > > _______________________________________________ > 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 Ciprus > Integration engineer > http://www.acision.com > > 9954 Mayland Dr > Suite 3100 > Richmond, VA 23233 > USA > T: +1 804 762 5601 > E: daniel.ciprus(a)acision.com > > ________________________________ > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you for > understanding. > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

TCP reset
by Daniel Ciprus 24 Apr '14

24 Apr '14

Hi, Is there any way to remove registration entry from registrar (Kamailio IMS built on stable 4.1) detected on p-cscf by incoming TCP RST from the network ? What's happening is that client is loosing connections/crashing and TCP RST is sent back to pcscf which is not propagating changes back to registrar. This is likely not possible with my version of kamailio but at least logging which would indicate change of the status will be helpful. thanks for any hints -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 3

Planning release of v4.1.3
by Daniel-Constantin Mierla 24 Apr '14

24 Apr '14

Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 8

Difference between route(RELAY) and t_relay
by aawaise 24 Apr '14

24 Apr '14

Hi all, I want to know while coding kamailio.cfg. What factor determines that route(RELAY) should be used or t_relay ?? Thanks. Cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Difference-between-route-RELAY-and-… Sent from the Users mailing list archive at Nabble.com.

2 1

LDAP and Kerberos backend for authentification / Or PAM / SASL ?
by Yoann Gini 23 Apr '14

23 Apr '14

Hello, My users are managed on a common LDAP / Kerberos setup. I don’t have access in any ways to the user password in clear text mode. I’ve start to follow this tutorial http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth and check this documenation http://kamailio.org/docs/modules/4.1.x/modules/ldap.html But it seems LDAP module provided in Kamailio don’t handle this scenario, it except to have user password in clear text mode in the LDAP database (wrong in so many ways…). Do we have a way with existing Kamailio plugin to use a authentication backend who don’t provide access to clear text password? I provide different authentication « adapter » who can help any service not compatible with Kerberos to handle authentication: - ldap bind - sasl - pam Do we’ve a way to use one of this backend as an authentication service (and not just a users database). Best regards, Yoann

2 4

IBM SIP server has implemented proxy managed overload protection
by Yang Hong 23 Apr '14

23 Apr '14

Hello Folks. According the new update on Feb 5, 2014, IBM has implemented proxy managed overload protection (PMOP) at SIP proxy server. http://pic.dhe.ibm.com/infocenter/wasinfo/v7r0/index.jsp?topic=%2Fcom.ibm.w… The above IBM link lists IBM SIP proxy server custom properties for SIP overload protection/control. •burstResetFactor •deflatorRatio •dropOverloadPackets •inDialogAveragingPeriod •maxThroughputFactor •outDialogAveragingPeriod •perSecondBurstFactor •proxyTransitionPeriod •sipProxyStartupDelay IBM SIP overload protection mechanism can be regarded as "Local SIP Overload Protection/Control" (Proceedings of WWIC, June 2013). I am moderator of open-source FreeRDP-WebConnect project at GitHub. https://github.com/FreeRDP/FreeRDP-WebConnect/issues/36 I will try to integrate two implicit SIP overload protection/control algorithms (RRRC, IEEE Globecom 2010 and RTDC, IEEE ICC 2011) into Kamailio (OpenSER) in the future. A survey on SIP overload protection/control algorithms (including IETF RFC "SIP Overload Control") can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, “A Comparative Study of SIP Overload Control Algorithms,"Network and Traffic Engineering in Emerging Distributed Computing Applications, Edited by J. Abawajy, M. Pathan, M. Rahman, A.K. Pathan, and M.M. Deris, IGI Global, 2012, pp. 1-20. http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SI… http://arxiv.org/abs/1210.1505 This survey paper provides a short review on SIP Express Router (SER). As we know, SIP Express Router (SER) and Kamailio (OpenSER) are open-source SIP router. "SIP Express Router (SER) provides a load balancing module to mitigate the overload caused by large subscriber populations or abnormal operational conditions (IP Telecommunications Portal, 2011). http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67… The presentation slides for two implicit SIP overload protection/control algorithms (RRRC and RTDC) are available for your download. Redundant Retransmission Ratio Control (RRRC) - implicit SIP overload protection/control algorithm (IEEE Globecom 2010 Slides) can be downloaded from the following ResearchGate link. https://www.researchgate.net/publication/258555827_Mitigating_SIP_Overload_… Round-Trip Delay Control (RTDC) - implicit SIP overload protection/control algorithm (IEEE ICC 2011 Slides) can be downloaded from the following ResearchGate link. https://www.researchgate.net/publication/257945199_Round-Trip_Delay_Control… The paper with Redundant Retransmission Ratio Control (RRRC, implicit SIP overload protection/control) algorithm can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, "Mitigating SIP Overload Using a Control-Theoretic Approach," Proceedings of IEEE Globecom, Miami, FL, U.S.A, December 2010. https://www.researchgate.net/publication/221284946_Miigating_SIP_Overload_U… http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5683124 Redundant Retransmission Ratio Control (RRRC, implicit SIP overload protection/control) algorithm has been quickly adopted by The Central Weather Bureau of Taiwan for their early earthquake warning system. Ting-Yun Chi, Chun-Hao Chen, Han-Chieh Chao, and Sy-Yen Kuo, "An Efficient Earthquake Early Warning Message Delivery Algorithm Using an in Time Control-Theoretic Approach", 2011. http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15# http://www.ipv6.org.tw/docu/elearning8_2011/1010004798p_3-7.pdf Short review and comments on RRRC implicit SIP overload protection/control algorithm by former IEEE TAC Associate Editor S. Mascolo: Luca De Cicco, Giuseppe Cofano, and Saverio Mascolo, "Local SIP Overload Control", Proceedings of WWIC, June 2013. http://link.springer.com/chapter/10.1007%2F978-3-642-38401-1_16# http://c3lab.poliba.it/images/2/2a/SipOverload_WWIC13.pdf The paper with Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) algorithm can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, "Design Of A PI Rate Controller For Mitigating SIP Overload," Proceedings of IEEE ICC, Kyoto, Japan, June 2011. https://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Cont… http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5963029 Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) algorithm has been recommended as White Paper by TechRepublic (CBS Interactive) http://www.techrepublic.com/whitepapers/design-of-a-pi-rate-controller-for-… Control theoretic approaches have been applied to model the interactions between an overloaded SIP server and its upstream servers as a feedback control system in two different scenarios - Round-Trip Delay Control (IEEE ICC 2011) and Redundant Retransmission Ratio Control (IEEE Globecom 2010). Journal paper (implicit SIP Overload Protection/Control) published by Springer Telecommunication Systems not only conducts more theoretical analysis of Round-Trip Delay Control (RTDC) and Redundant Retransmission Ratio Control (RRRC), but also discusses how to apply RTDC algorithm to mitigate SIP overload for both SIP over UDP and SIP over TCP (with TLS). Journal paper can be downloaded from the following ResearchGate link. Y. Hong, C. Huang, and J. Yan, "Applying control theoretic approach to mitigate SIP overload", Telecommunication Systems, Volume 54, Issue 4, December 2013, pp 387-404. https://www.researchgate.net/publication/257667871_Applying_control_theoret… http://link.springer.com/article/10.1007/s11235-013-9744-8 Implicit SIP overload protection/control solution using control theoretic approaches can be regarded as an engineering applications of control theory, see the discussion on control system design in the answers to the following two ResearchGate(RG) questions. ResearchGate is the largest social network for research scientists and engineers in the world. RG Question #1: "What are trends in control theory and its applications in physical systems (from a research point of view)?" https://www.researchgate.net/post/What_are_trends_in_control_theory_and_its… RG Question #2: "What are the latest problems to be solved in "control of nonlinear systems"?" https://www.researchgate.net/post/What_are_the_latest_problems_to_be_solved… RG Question #3: "What are the new trends in control engineering?" https://www.researchgate.net/post/What_are_the_new_trends_in_control_engine… Best regards, Winston Hong Software Engineer Ottawa, Ontario Canada

1 0

rtpproxy_manage() issues in rtpproxy-ng
by Alex Balashov 23 Apr '14

23 Apr '14

Hello, I'm running the latest pull of Kamailio 4.1 (last commit be187e135b0b9b28136817c3569ab5c0abcc5b3f) and am using rtpproxy-ng with a recent mediaproxy-ng master (commit cb6990e43864b077dd6a24acfbdf5ef76c1a427e). For no apparent reason, Kamailio has stopped sending 'offer' commands to it when I use rtpproxy_manage(). My use of it is in this setting: if(isflagset(PROXY_MEDIA) && !isflagset(PROXY_MEDIA_SET) && has_body("application/sdp")) { set_rtp_proxy_set("1"); rtpproxy_manage("o"); add_rr_param(";proxy_media=yes"); setflag(PROXY_MEDIA_SET); } I put in a log message to confirm that this block is being run. It is. But, there's no offer command going to the mediaproxy-ng on the other box, as confirmed by packet captures and logs. However, the mediaproxy-ng is getting lots of 'answer' and 'delete' commands that it doesn't know what to do with, since the call was never initialised with 'offer'. When I change rtpproxy_manage() to rtpproxy_offer(), it works perfectly. I've been using rtpproxy_manage() forever, almost as long as it's existed. Does it not work with rtpproxy-ng, despite what the documentation says? -- 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 8

RTPProxy and premature RTP
by Daniel Tryba 23 Apr '14

23 Apr '14

I'm having some troubles with a provider sending RTP before a 183 Session Progress or 200 OK (I see up to 1s of rtp prematurely). The machine is running rtpproxy and apparently rtpproxy buffers these rtp packets and flushed them in one burst when the 183/200 arrive, this creates havock in some endpoints but is undesirable in all cases IMHO if the delay of the 182/200 is to high (>0.1s). Is there a way to control buffering/flushing from kamailio? Is rebuilding rtpproxy with a smaller buffer an option? Or should I switch to an other proxy module? Any thoughts about this subject (apart from getting the provider to stop sending premature rtp)? -- 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

Kamailio-Asterisk - route "FROMASTERISK" not found
by Patrik Kristel 23 Apr '14

23 Apr '14

Hello all, I'm trying to implement Kamailio 4.1 with Asterisk 12.1.0 regarging this tutorial: > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And when I try to compile kamaili.cfg, I still got this error: ERROR route "FROMASTERISK" not found I have loaded all modules like in tutorial. I tried to find some solve this issue, but with no result.

2 2

Re: [SR-Users] Drouting
by Keith 23 Apr '14

23 Apr '14

Hi, Or is it possible to use a regular expression for the user? I basically want certain users to use a certain route without having to add them all in. Keith

1 0

Loadbalancer/dispatcher with dialog/cnxcc
by Oliver Roth 23 Apr '14

23 Apr '14

Hi all Following situation 1 dispatcher/loadbalancer getting all the inbound traffic and sending it to 3 different gateway (round robin). The loadbalancer has no (or even very few) business logic. Just "in" - split to different gateway - "out" 3 sip gateway doing all the business logic like auth, modifications on header, different checks, ... These 3 gateways use the same database (cluster) with routing tables and so on. A call gets terminated to a carrier through these sip gateway. Now I would like to implement call limiting (no of calls) by using either only dialog module or cnxcc module based on "source ip" or later on "cli". My problem: A call from one source ip can be sent to the (3) different sip gateway - so not all calls are processed by the same sip gateway. How can I ensure, that only a certain number of calls are allowed - even if they are split up on the 3 different gateway? Or do I need to implement this kind of business logic on the dispatcher/loadbalancer? That would not make much sense, because this is just a "stupid" machine... Thanks for helping Oli

1 0

Pick domain name of a user from location list of siremis
by aawaise 23 Apr '14

23 Apr '14

Hello all, I want to pick the domain name of a particular online user from location list of siremis and then perform some action on this basis. How can it be picked ?? Secondly how can the user name of called user be picked from the INVITE packet received by the server ?? Any help will be appreciated. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Pick-domain-name-of-a-user-from-loc… Sent from the Users mailing list archive at Nabble.com.

1 0

Drouting
by Keith 22 Apr '14

22 Apr '14

Hi, I am trying to use the drouting module and was wondering if there is a wildcard character for the dr_groups table for the user? My situation is that I want to look at any user from a certain IP range (i.e. the carrier). Any ideas? Thanks Keith

2 1

Issues with Kamailio Asterisk Integracion
by William Baylón Huerta 22 Apr '14

22 Apr '14

Hi, i see this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb, I've done step by step as described in the tutorial, but i have a problem, when A extension dial to B extension, B extension doesn't ringing... I send my sip.conf and kamailio.cfg Addittional, asterisk and kamailio are installed on same server, which have one private IP (192.168.50.217) and public IP 200.41.110.94 -- *Kevin W. Baylón Huerta* *Departamento Sistemas - Sitatel.* *Teléfono**: (51) 17073501 Anexo 208* *Celular: (51) 989715598*

2 1

SIREMIS - SQL-based CDR rating engine
by Dea Putera 22 Apr '14

22 Apr '14

Hi, I'm looking for any documentation on SIREMIS especially related to the rating engine. Where can I find it ? Or at least which module(s) are related to the rating engine. Any help is greatly appreciated. Thanks. -- -Dea

2 1

Caller ID number being passed to Gateway
by Joel E White 21 Apr '14

21 Apr '14

I have been playing with this for a couple days. What I have pulls from the usr_preferences DB and "Should" insert CID Name and CID Number What I am ending up with is that the name portion is being sent, not the number. Is there a certain format that I should add the DID in usr_preferences? Also when I changed the AVP type in the DB from a 0 to a 1 the value for CID DID changes... can someone help me with correcting this behavior? avp_db_load("$from","$avp(s:callerid-name)"); avp_db_load("$from","$avp(s:caller-did)"); uac_replace_from("$avp(s:callerid-name)","$avp(s:callerid-did)"); xlog("SIP From Header returned $hdr(From)\n"); xlog("AVP CallerID-Name returned $avp(s:callerid-name)\n"); xlog("AVP CallerID-Ext returned $avp(s:callerid-did)\n"); Thank you, Joel --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

4 10

tls.so, no module matching <tls> found on FreeBSD 10
by Yoann Gini 21 Apr '14

21 Apr '14

Hello, It’s my first message here so here is a little introduction about me. I’m Yoann gini, a french system administrator specialized in Apple product for enterprise. I also manage some other UNIX sever (essentially BSD based) and some windows when customers pay me a lot… I’m trying to install Kamailio on by personal FreeBSD 10 server. My intend is discover products functionalities and see what can be done whit it. Maybe I can made a package for OS X Server and deploy it along with e-mail and xmpp services when my customers need efficient communication system. So, at this time, I’m playing with various options and I’ve some problems. The one who lead me here concern TLS module. I’m unable to start the service with TLS module enabled, digging into the debug log, I’ve catch those lines: 0(35943) DEBUG: <core> [cfg.y:1635]: yyparse(): loading module tls.so 0(35943) DEBUG: <core> [sr_module.c:591]: load_module(): load_module: trying to load </usr/local/lib64/kamailio/modules/tls.so> 0(35943) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... 0(35943) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <tls> found It seems that the module is loaded, but impossible to configure… Here is the output of grep -i tls kamailio.cfg ### #!define WITH_TLS # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS #!ifdef WITH_TLS listen=tls:<ipv4>:5061 listen=tls:<ipv6>:5061 enable_tls=yes #!ifdef WITH_TLS loadmodule "tls.so" #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") And the content of /usr/local/etc/kamailio/tls.cfg [server:default] method = TLSv1 verify_certificate = yes require_certificate = no private_key = /etc/ssl/keys/my_real_ssl_cert.key certificate = /etc/ssl/certs/my_real_ssl_cert.crt ca_list = /etc/ssl/cert.pem #crl = /usr/local/etc/kamailio/crl.pem [client:default] verify_certificate = no require_certificate = no After some search, I’ve found this message on the list http://lists.sip-router.org/pipermail/sr-users/2013-August/079439.html It seems that a bug on OpenSSL side cause some trouble with TLS module… Is it still true ? How do you manage your TLS configuration ? I’m using OpenSSL 1.0.1g with kamailio 4.1.2. Best regards, Yoann.

2 2

Pick Domain name from Location Table
by aawaise 21 Apr '14

21 Apr '14

Hi All, When my kamailio server receives an "invite" packet to make a call. I want the kamailio server to make a decision about call routing. Whether to route call to other kamailio server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. I have two tasks here. One is to pick the domain of user from "Location Table" of Siremis. And secondly, route this call to other server on basis of the string I get from "Domain" of "Location Table". Need help regarding changes in kamailio.cfg for this. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Pick-Domain-name-from-Location-Tabl… Sent from the Users mailing list archive at Nabble.com.

2 1

PCSCF configuration towards pcrf
by Dees 19 Apr '14

19 Apr '14

Hi, Very new to kamailio, following instruction have copied the examples file for pcscf and ran kamaillo. I have installed ims modules, but I don't see to see any connection initiation towards to PCRF? Any clues on what I am doing wrong. Thanks. Dees root@kamalio:/etc/bind# more /etc/hosts 127.0.0.1 localhost 10.209.96.85 pcscf.kamailio-ims.org kamalio #10.209.96.92 pcrf.kamailio-ims.org root@kamalio:/etc/bind# ping pcrf.kamailio-ims.org PING pcrf.kamailio-ims.org (10.209.96.92) 56(84) bytes of data. 64 bytes from 10.209.96.92: icmp_req=1 ttl=64 time=0.392 ms

1 3

event_route[tm:branch-failure] question
by Juha Heinanen 18 Apr '14

18 Apr '14

in branch route i execute: t_on_branch_failure("contact"); xlog("L_INFO", "Routing $rm <$ru> to contact\n"); and in branch_failure route i have: event_route [tm:branch-failure:contact] { if (t_check_status("486")) { xlog("L_INFO", "Got 486 response to <$ru>\n"); }; and i get to syslog: Apr 13 21:44:07 siika /usr/sbin/sip-proxy[17475]: INFO: Routing INVITE <sip:test-0x11107b0@192.98.102.30:5064> to contact Apr 13 21:44:11 siika /usr/sbin/sip-proxy[17437]: INFO: Got 486 response to <sip:test@test.tutpro.com> why has $ru in branch_failure route changed from $ru of the branch to original request uri (aor)? i would expect to see in $ru of branch_failure route the $ru of the branch. is this a bug? if not, is there any means in branch_failure route to get access to the branch that failed. -- juha

4 34

kamailio db
by Slava Bendersky 18 Apr '14

18 Apr '14

Hello Everyone, Is possible run kamailio without database ? I got ldap working and extension is registered, but can't place calls between two extensions. Getting 404 all the time. Slava.

4 43

invite always contains "s"
by Vasco Mendes 18 Apr '14

18 Apr '14

Hi, I created an incoming trunk between iptel and my asterisk. Whatever the account is on iptel I always have in the from of received INVITE < sip :s@mypublicip :5060 >. < s > is not a recognized DID on my asterisk. Why do I receive this < s > in the from ?? Many thanks for your help, Vasco --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 1

Editings in kamailio.cfg for routing decision
by aawaise 18 Apr '14

18 Apr '14

Hi All, When a user sends an "invite" packet to make a call. I want to make a decision about call routing. Whether to route call to other server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. What changes are to be made in kamailio.cfg ?? Actually in my environment my user can connect to any of the servers. I want to know to which server it is connected and then make the call routing decision. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Editings-in-kamailio-cfg-for-routin… Sent from the Users mailing list archive at Nabble.com.

1 0

Custom fr timer for dialog's OPTIONS keep alives
by Vassilis Radis 17 Apr '14

17 Apr '14

Hello, I would like to try customizing the fr_timer for the OPTIONS keep-alive msgs transactions that the dialog module sends. I tried to call t_set_fr() function from inside the dialog modules's code (in dlg_req_within.c , in dlg_send_ka() function after the call to d_tmb. t_request_within(&uac_r); But it seems that this tm function is not available from inside other module's code. Is the above is the correct place to try and do it, and how can I programmatically ,inside a module's code, adjust fr_timer? On a side note, i followed the call flow from dlg_send_ka() down to the t_uac_prepare() function of tm module, where the fr_timer is set. In there, there is this comment+code: /* init timers hack, new_cell->fr_timer and new_cell->fr_inv_timer * must be set, or else the fr will happen immediately * we can't call init_new_t() because we don't have a sip msg * => we'll ignore t_set_fr() or avp timer value and will use directly the * module params fr_inv_timer and fr_timer -- andrei */ new_cell->fr_timeout=cfg_get(tm, tm_cfg, fr_timeout); new_cell->fr_inv_timeout=cfg_get(tm, tm_cfg, fr_inv_timeout); new_cell->end_of_life=get_ticks_raw()+lifetime; Does this mean that it cant be done in a straighforward way? Any advice on how I can achieve the customized fr_timers for those messages? Thanks.

2 2

Basic question regarding Routing
by aawaise 17 Apr '14

17 Apr '14

Hello, I want to ask some basic questions. Anyone please guide me to links with answers. I am sure such basics must be explained online. 1. What is exactly achieved by "route(RELAY)" ?? Routing calls from local users to other domains ?? 2. If user 1 is caller and user 2 is being called. Which pseudo variable will be used to catch value of called user i.e user 2 and his domain ?? I want to pick domain name from location table in database and not from the authorization module. Thanks, Cheers, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Basic-question-regarding-Routing-tp… Sent from the Users mailing list archive at Nabble.com.

2 1

Routing decision based on domain entry in Location Table
by aawaise 17 Apr '14

17 Apr '14

Hello, I have deployed two kamailio servers, both sharing the same database. I want my kamailio servers to make a decision on the routing of incoming call by seeing the domain of "called user" from location table of online users. If the "called user" is connected to same server, the call should be routed to user directly. In other case if the user is connected to other server, the call should be forwarded to other user for further processing. Will someone please guide me how can this be achieved ?? P.S : I am using kamailio-3.1.1. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Routing-decision-based-on-domain-en… Sent from the Users mailing list archive at Nabble.com.

2 4

rls multipart body does not contain watcherinfo+xml
by Mihai Marin 16 Apr '14

16 Apr '14

Hello Sirs, I'm trying to accomplish a rls subscribe but I have some miss-understandings regarding the multipart publish response that should come. I'm using the integrated xcap from the kamailio with all the xml files added. My problem is that event if I request a multipart containing application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml I only get the first two. The request contains the following headers: Event: presence Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml The response only sends the first two: --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:bob@93.187.138.214.681064155> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri="sip:bob@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="4" fullState="false"><resource uri="sip:alice@93.187.138.214"><instance id="Scf8UhwQ" state="active" cid="1397508890.sip:alice@93.187.138.214.1552580371"/></resource></list> --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:alice@93.187.138.214.1552580371> Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8" standalone="no"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:alice@93.187.138.214"> <dm:person id="p6850"> <rpid:activities/> <rpid:status-icon>http://93.187.138.214:5060/xcap-root/oma_status-icon/users/sip:alice@93.187…</rpid:status-icon> </dm:person> <tuple id="t7673"> <status> <basic>open</basic> </status> <contact>sip:alice@93.187.138.214</contact> <note>Online</note> </tuple> </presence> --4kSPpRGyJQct4DIhyGnyobqO-- What do I missing? I need the body part with the watchers info in oder to see the state of the watcher (active, pending) - to accomplish the authorization for users. Thank you in advance for you help. Best regards, Mihai M

2 3

Register request forwarding from kamailio to asterisk
by betul.cerit＠netgsm.com.tr 16 Apr '14

16 Apr '14

Hi to all We integrated kamailio and asterisk as explained here http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb When kamailio receives a register request it authenticates and forwards the request to asterisk.Why do we need to do this?What happens if we dont pass registration requests to asterisk? Thanks in Advance Betül

2 2

Security
by Keith 16 Apr '14

16 Apr '14

Hi All, I am looking to apply some security on invites so I only accept from IP addresses in the address table or authenticated users. I have the address table bit working but I am struggling with the authenticated users bit. Wondering if anyone has done it? I was thinking of using the location table in some way as those users have authenticated? Any ideas would be appreciated. Keith

2 3

uac_auth problem
by neumann 16 Apr '14

16 Apr '14

Hi all sorry for my english! Iam trying to authenticate as uac on 3rd party server(Asterisk). Kamailio get 401, send next invite with DIGEST but Asterisk send 401 again((( Asterisk don’t write in log what auth is failed. 115.600045 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1093 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.601818 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.602470 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx 115.604310 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1267 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.605537 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.606431 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx My aster peer: [kamailio] type=peer defaultuser=kamailio secret=1234 host=dynamic disallow=all allow=alaw nat=no qualify=yes My kamailio config: modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") route[DISPATCH_OUT] { if(!ds_select_dst( 4, 8)){ send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); t_relay(); exit; } failure_route[RTF_DISPATCH_OUT] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(s:uac_user) = "kamailio"; $avp(s:uac_pass) = "1234"; $avp(s:uac_realm) = "asterisk"; if (isflagset(7)) { t_reply("513","Authentication failed"); xlog("L_INFO", "[%ci]: Remote authentication failed\n"); break; } if (uac_auth()) { xlog("L_INFO", "401/407 message recived"); # mark that auth was performed setflag(7); # trigger again the failure route t_on_failure("RTF_DISPATCH_OUT"); # repeat the request with auth response this time append_branch(); #t_relay_to_udp("94.25.100.198","5060"); t_relay(); break; } else { xlog("L_INFO", "uac_auth filed!!!!!!!!!!!"); } t_reply("514","Unknow authentication peer"); xlog("L_INFO", "[%ci]: Unknown authentication peer.\n"); break; } # next DST - only for 500 or local timeout if (t_check_status("500") or t_check_status("480") or (t_branch_timeout() and !t_branch_replied())){ ds_mark_dst("p"); if(ds_next_dst()){ t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); route(RELAY); exit; } } } ———————————— Timofeev Dmitry VoIP Engineer Linux, Asterisk, Freeswitch, Cisco solutions Skype: itsroot icq: 227227933

2 1

client provisioning.
by Slava Bendersky 15 Apr '14

15 Apr '14

Hello Everyone, Is possible do phone provisioning with kamailio ? Slava.

3 8

PERMISSIONS module issue
by Samuel Ware 15 Apr '14

15 Apr '14

I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show displays the new address; however kamcmd permissions.addressDump does not neither does kamcmd permissions.subnetDump. The messages from this new address return a false to the !allow_source_address("1”) command in my routing logic. I am wondering if this is a bug or I am doing something wrong. I am on the most recent GIT version to the best of my knowledge. Sam

6 6

SDP body rewritten issue at RTPproxy
by Wingsravi R 15 Apr '14

15 Apr '14

Dear All, I am working with Kamailio (4.1.2) and RTPproxy (1.2.1) servers and Jitsi as SIP clients. I have a problem in Audio/Video calling with Kamailio+RTPproxy server combination that Video is not streaming to other side through RTPproxy(also no audio both sides), I am suspecting something is rewritten in SDP body at RTPproxy, concerning to audio/video parameters (I am unable to rectify what is causing problem there) . Please find the Tcpdump based SIP capture of video call with this Kamailio+RTPproxy set-up. And with only kamailio server running (without RTPproxy instance running), audio/video call is fine on both sides. (find SIP capture for the same call). Also find my kamailio.cfg file for any reference. Please anybody help me in rectifying problems at RTPproxy server. Awaiting somebody's comments on this issue. Regards, Ravi

3 2

Re: [SR-Users] SDP c= IP doesn't get rewritten
by Ciprus, Daniel 14 Apr '14

14 Apr '14

Yes I do Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Daniel-Constantin Mierla Date:04/11/2014 6:13 AM (GMT-05:00) To: "Ciprus, Daniel" ,"Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] SDP c= IP doesn't get rewritten Hello, are you using a rtp relay (e.g., rtpproxy)? Cheers, Daniel On 11/04/14 02:46, Daniel Ciprus wrote: All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, 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 GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding. _______________________________________________ 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 ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

4 5

LUA - UAC package implementation
by Daniel Ciprus 11 Apr '14

11 Apr '14

Hi, How much of UAC package is implemented in LUA embedded module ? According to wiki seems like only "sr.uac.replace_from" is implemented. My goal is to "emulate" 3rd party registration by using UAC and LUA meaning that I will have to recreate register, tweak existing attributes and send it over to application server to process. thanks Dan -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

3 4

about tls client certificates
by Juha Heinanen 11 Apr '14

11 Apr '14

while doing some tls tests, i noticed that if tls.cfg has a section like this [server:default] verify_certificate = no require_certificate = no tls_method = SSLv23 private_key = /etc/sip-proxy/certs/sip-proxy/key.pem certificate = /etc/sip-proxy/certs/sip-proxy/cert.pem ca_list = /etc/ssl/certs/cacert.org.pem then client does not give its certificate to kamailio server during tls connection setup even if it had one. if i specify: require_certificate = no then client sends its certificate to kamailio server, but if another client does not have a client certificate, then it cannot connect at all. one way to solve this would be making kamailio listen on two tls ports, one for clients that are required to present a a certificate and another port for clients that do not have a certificate. unfortunately, it is not possible to add a mask to ip address in tls.cfg section like this: [server:0.0.0.0/0:5062] does anyone have a solution to this problem (other that running two kamailio instances)? -- juha

2 2

SDP c= IP doesn't get rewritten
by Daniel Ciprus 11 Apr '14

11 Apr '14

All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, 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 GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 1

Disjoint REGISTER and INVITE username/uri sets
by James Cloos 10 Apr '14

10 Apr '14

Given a proxy where the only registrations will be next-hop pbxes, where those names will not be INVITEable, what is the ideal way to tell kama the list of accepted usernames and authentication info for REGISTER? It is important to ensure that any INVITE to any of the REGISTERable usernames 404s. Thanks, -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 1024D/ED7DAEA6

3 3

carrierroute prefix regex?
by Igor Potjevlesch 10 Apr '14

10 Apr '14

Hello, I'm wondering if we can set into < scan_prefix > an entrier with regex like : 3[0-9]{3}? If not, what would be the best module to manage in a DB the prefix that can be composed by SIP clients before call cr_route? Something equivalent to a dialplan I guess. Thanks for your feedback. Regards, Igor.

1 1

Re: [SR-Users] How Do you Stack up Against Cisco's Call Manager
by Marjorie Lawrence 10 Apr '14

10 Apr '14

Hi, Not sure if this is a fair question. But how do you compare your call processing engine against Cisco's Call Manager? Appreciate your time. Marjorie Lawrence Global Wireless Networks, Inc. Senior Voice Engineer/Unified Communications Email: <mailto:Marjorie@gwnwireless.com> Marjorie(a)gwnwireless.com Website: <http://www.gwnwireless.com/> http://www.gwnwireless.com Office (301)674-5933

1 0

how to track down issue with shared lines
by mark li 10 Apr '14

10 Apr '14

I have a scenario where calls that are transferred to a shared line doesn't trigger the "in use" light to show up on both phones that have the shared line. but when i make an outgoing call, let's say, using the shared line, the "in use" light lights up on both phones that have the shared line. Phone - polycom 650s. Just a noob with SIP, kamailio and telecom, so not sure where to start debugging. Any suggestions would be appreciated. thanks in advance for your time.

1 0

Load fr_inv_timer from DB
by Igor Potjevlesch 10 Apr '14

10 Apr '14

Hello, I try to set the "fr_inv_timer" dynamically from DB. Under 60s it works fine. But I can't set more than 60s. Here are the TM parameters: # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 60000) Here are the instructions to set dynamically: avp_db_load("$ruri/username", "$avp(s:inv_timeout)/usr_timeout"); t_set_fr($avp(s:inv_timeout)*1000); Is there anything wrong? Regards, Igor.

1 0

topoh and onhold reinvites
by Daniel Tryba 10 Apr '14

10 Apr '14

I'm having some trouble with kamailio that uses the topology hider module and an incoming call being put onhold by the caller. The INVITE containing the sdp a=sendonly is to the 172.19.162.1 (the topoh mask_ip) contact header from the 200 OK to the initial INVITE. U 109.235.36.178:5060 -> 109.235.32.40:5060 INVITE sip:172.19.162.1:5060;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone SIP/2.0. This is handled within dialog (has_totag() && !loose_route()), which in the default config ends in a sl_send_reply("404","Not here") Explicitly handing INVITE (just relaying), results in the INVITE being sent to 172.19.162.1 (which doesn't exist) U 109.235.32.40:5060 -> 172.19.162.1:5060 INVITE sip:172.19.162.1:5060;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone SIP/2.0. How to get this INVITE to the actual destination? To get the same result for the INVITE as for the ACK to the 200 OK? U 109.235.32.40:5060 -> 109.235.36.178:5060 SIP/2.0 200 OK. ... Contact: <sip:172.19.162.1;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*>. U 109.235.36.178:5060 -> 109.235.32.40:5060 ACK sip:172.19.162.1;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy* SIP/2.0. U 109.235.32.40:5060 -> 109.235.34.107:5060 ACK sip:+31756418030@109.235.34.107:5060;transport=udp SIP/2.0. -- 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 3

ldap SSL
by Slava Bendersky 10 Apr '14

10 Apr '14

Hello Everyone, How I can specify ldap CA cert in ldap configuration ? The directive ca_list bellow is ignored. [07/Apr/2014:13:40:37 -0400] conn=20 op=-1 fd=70 closed - Peer does not recognize and trust the CA that issued your certificate. [07/Apr/2014:13:40:37 -0400] conn=22 op=-1 fd=64 closed - Encountered end of file. Current config ldap_server_url = "ldaps://ds389.network.com" ldap_bind_dn = "uid=adm,ou=People,dc=network,dc=com" ldap_bind_password = "password" authtype = simple tls = on ca_list = /etc/kamailio/cert/ca-cert_ldap.pem

2 3

$sel(tls.peer.subject.cn) error
by Juha Heinanen 10 Apr '14

10 Apr '14

i have these two lines in config: $var(common_name) = @tls.peer.subject.cn; $var(common_name) = $sel(tls.peer.subject.cn); the latter gives error at startup: 0(24214) ERROR: <core> [select.c:316]: resolve_select(): Unable to resolve select 'tls' at level 0 0(24214) ERROR: <core> [select.c:177]: w_parse_select(): parse_select: error while resolve_select 'tls.peer.subject.cn' 0(24214) ERROR: pv [pv_select.c:45]: pv_parse_select_name(): invalid select name [tls.peer.subject.cn] 0(24214) ERROR: <core> [pvapi.c:839]: pv_parse_spec2(): pvar "sel" has an invalid name param [tls.peer.subject.cn] 0(24214) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [)/41] in [$sel(tls.peer.subject.cn)] at [24 (5)] 0(24214) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 874, column 29-53: Can't get from cache: $sel(tls.peer.subject.cn) ERROR: bad config file (1 errors) based on pseudo variable wiki page, the syntax should be correct. what is wrong with $sel(tls.peer.subject.cn)? -- juha

2 5

Re: [SR-Users] LCR
by Keith 10 Apr '14

10 Apr '14

Hi, I am having issues with trying to use multiple lcr_ids so I can route to different groups. When I reload lcr it always only shows one group. I have copied the first group exactly and just changed the number but can't work out why it's not working. Any ideas? Thanks Keith

2 1

Exported params of USRLOC module
by aawaise 10 Apr '14

10 Apr '14

Hello, I am using kamailio-3.1 with siremis. I want to ask some questions about the exported params of usrloc module defined in the link given below. http://kamailio.org/docs/modules/3.1.x/modules_k/usrloc.html#id2997066 1. These exported params export value to whom ?? Other params or siremis location table ?? 2. If any export param is not defined in kamailio.cfg. Does its default value considered automatically or it is not used at all ?? 3. In siremis location table, I am not able to show my domain name of selected online user when I click on user. I want to do it. User's contact, received and other columns are filled but not domain name :( Shall I dig deep into usrloc module or I need to follow some other path to resolve this issue ?? Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Exported-params-of-USRLOC-module-tp… Sent from the Users mailing list archive at Nabble.com.

2 1

Show Domain-Name in Siremis
by aawaise 10 Apr '14

10 Apr '14

Hello, I am running kamailio server in my set up. And have got siremis working for monitoring purposes. Problem is that in web interface, domain name of users is not shown in siremis. If a user is online, I get following credentials infront of "Client" column, Client.Name@Client's.IP How can I show domain name in siremis ?? Any help will be highly appreciated. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp12660… Sent from the Users mailing list archive at Nabble.com.

2 7

Segfault when trying to send MESSAGE through websocket with jsSIP
by Alex Villacís Lasso 09 Apr '14

09 Apr '14

I am currently trying to replace a pure-Asterisk implementation of SIP messaging through Websockets, with a Kamailio-4.1.2-based implementation. However, when I try to send a message with jsSIP, Kamailio crashes: Program terminated with signal 11, Segmentation fault. #0 0x00007f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 350 slot = reg_get_entry(hash, _reg_htable->htsize); Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.132.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.10.3-10.el6_4.6.x86_64 libcom_err-1.41.12-18.el6.x86_64 libgcc-4.4.7-4.el6.x86_64 libselinux-2.0.94-5.3.el6_4.1.x86_64 libtool-ltdl-2.2.6-15.5.el6.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-14.el6.x86_64 mysql-connector-odbc-5.1.5r1144-7.el6.x86_64 mysql-libs-5.1.73-3.el6_5.x86_64 nss-softokn-freebl-3.14.3-9.el6.x86_64 openssl-1.0.1e-16.el6_5.4.x86_64 unixODBC-2.2.14-12.el6_3.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 #1 0x00007f0e5cf36c71 in uac_reg_lookup (msg=0x7f0e6271e790, src=0x7fff59734b00, dst=0x7f0e6267c950, mode=0) at uac_reg.c:924 #2 0x00007f0e5cf2f991 in w_uac_reg_lookup (msg=0x7f0e6271e790, src=0x7f0e6267e0d0 "\a", dst=0x7f0e6267c950 "\006") at uac.c:560 #3 0x0000000000419bf6 in do_action (h=0x7fff59735690, a=0x7f0e6267f1e0, msg=0x7f0e6271e790) at action.c:1111 #4 0x0000000000422878 in run_actions (h=0x7fff59735690, a=0x7f0e6267e510, msg=0x7f0e6271e790) at action.c:1599 #5 0x0000000000417900 in do_action (h=0x7fff59735690, a=0x7f0e62664aa0, msg=0x7f0e6271e790) at action.c:715 #6 0x0000000000422878 in run_actions (h=0x7fff59735690, a=0x7f0e6265d3b8, msg=0x7f0e6271e790) at action.c:1599 #7 0x0000000000423017 in run_top_route (a=0x7f0e6265d3b8, msg=0x7f0e6271e790, c=0x0) at action.c:1685 #8 0x00000000004a5153 in receive_msg ( buf=0x7f0e570d0168 "MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@gatitos."..., len=585, rcv_info=0x7f0e570cfe90) at receive.c:212 #9 0x00007f0e5c8e5802 in ws_frame_receive (data=0x7fff59735a40) at ws_frame.c:652 #10 0x000000000045531c in sr_event_exec (type=10, data=0x7fff59735a40) at events.c:254 #11 0x000000000052e04e in ws_process_msg ( tcpbuf=0x7f0e570d0160 "\201\376\002I\032\327\302\344MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@"..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1146 #12 0x000000000052e21f in receive_tcp_msg ( tcpbuf=0x7f0e570d0160 "\201\376\002I\032\327\302\344MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@"..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1182 #13 0x000000000052f2f3 in tcp_read_req (con=0x7f0e570cfe78, bytes_read=0x7fff59735cac, read_flags=0x7fff59735ca4) at tcp_read.c:1383 #14 0x0000000000530d27 in handle_io (fm=0x7f0e62724d30, events=1, idx=-1) at tcp_read.c:1617 #15 0x00000000005296c0 in io_wait_loop_epoll (h=0x8df220, t=2, repeat=0) at io_wait.h:1092 #16 0x0000000000531650 in tcp_receive_loop (unix_sock=53) at tcp_read.c:1728 #17 0x0000000000523c21 in tcp_init_children () at tcp_main.c:4959 #18 0x000000000046d6a3 in main_loop () at main.c:1702 #19 0x000000000047030b in main (argc=13, argv=0x7fff59736178) at main.c:2533 Is this a known bug?

2 8

message 484
by Slava Bendersky 09 Apr '14

09 Apr '14

Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ? <--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" <sip:1300@networklab.loc>;tag=as0a530a8d To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367(a)networklab.loc CSeq: 102 OPTIONS Contact: <sip:1300@192.168.100.145:5062>;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 Slava.

3 19

cnxcc
by Kelvin Chua 09 Apr '14

09 Apr '14

i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua

4 7

rfc: evapi architecture and message format
by Daniel-Constantin Mierla 09 Apr '14

09 Apr '14

Hello, (cross-posting because the results are impacting users) recently I added a new module named evapi: - http://www.kamailio.org/docs/modules/devel/modules/evapi.html The purpose is to able to send custom messages to applications that connect to kamailio via tcp on a special port (different than sip port) as well as receive messages and react on them. One of the first use case we look at is integration with cgrates for real time billing. For example: - send notification when a new call comes in and cgrates will return allow/reject - send notification when the call starts so cgrates can start billing - send notification when the call ends so cgrates will close bllling The module is now in quite good shape, sending and receiving messages. Sending is done from config file, in the future some modules might do it from inside. Received message is passed to config file via an event_route and variables. Now, the point of this discussion is to debate following points: 1) should there be a specific format used for this messages? The examples in the README use json. The module can encapsulate them in netstring format when sending over tcp to be easy to parse and read the message at once. This is optional, the message can be sent as it is 2) now messages are sent to all connected applications, should be there some subscription mechanism per event? If yes, then there has to be a way to specify the event, so point 1) might need a specific format 3) handle response - should be there a defined api that will say allow, reject, drop? I was thinking that the return could be some script for an embedded interpreter (e.g., Lua) that can be executed from config file. Or, alternative, manually scripted in the event route, so each user defines its own api 3) other suggestions for what should be discussed? Looking forward to opinions regarding these matters. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Kamailio IMS dnszone
by Andrea Meroni 09 Apr '14

09 Apr '14

Issue fixed: i just had to add the following line dns_try_naptr=yes to all the kamailio.cfg files of each of my IMS entity. I suggest to modify accordingly the config files in each of the examples (pcscf, icscf, scscf) in the master tree in order to make them properly working with the kamailio_ims_dnszone example Regards, Andrea

1 0

Dialog: Keep-alive issue:
by Vassilis Radis 08 Apr '14

08 Apr '14

Hello, I am trying to use the dialog module for dealing with some cases where clients do not send BYE messages correctly and just disappear. For this reason i use the ka_timer and ka_interval along with ka-dst and ka-src attributes. I am using the master branch. I have setup the following test: I have setup a sipp instance as a callee uas configured to discard OPTIONS messages. Kamailio serves as a proxy between this sipp instance and a VoIP device registered to kamailio. Kamailio is also configured to send every call to the sipp uas with stateful proxing and dialog support: CALLER ------> KAMAILIO -----> CALLEE (sipp / ignores OPTIONS msg) I have the following issue: 1. Dialog does send the OPTIONS messages to caller and callee, and as expected, only the caller responds (with a 200 OK response). But the dialog module never terminates the dialog as it should (because the callee never responds to the OPTIONS msg). Instead it keeps sending those OPTIONS for ever. It is like there is no timer entry in the timer list for those OPTIONS. I looked a bit in the source code and I am trying to find where a timeout for those keep alives is a)set and b)handled . I see that the dlg_timer_routine is called every second but after turning debugging on, i see that the only timer reported is the default timeout for the call which i ve set to 3600 secs in the module parameters. Any insight or advice? Thanks a lot.

2 3

uac_replace_from
by Kelvin Chua 08 Apr '14

08 Apr '14

Has anybody experienced having uac_replace_from() concatenate the old from: value after the new domain? for example: From: "user1" <user1(a)domain.com> becomes From: "user2" <user2(a)domain.comuser1> Kelvin Chua

5 7

Kamailio IMS dnszone
by Andrea Meroni 08 Apr '14

08 Apr '14

Hi all, i have a strange problem: i configured a basic IMS infrastructure Kamailio-based, with a DNS server (bind9) running on a virtual machine with IP address 192.168.100.55. Whenever i try to register a user, the first registration almost always fails, while the second is successful. I attached my dnszone config used with bind9 and the pcaps of the 2 regs; my questions are - is the IMS dnszone configuration correct? - the problems i got are just due to latency issues or have i misconfigured something? Regards, Andrea

1 0

RTP not coming via RTPProxy
by varun pratapsingh 08 Apr '14

08 Apr '14

HI All, I am using the Kamailio 4.1.2 with RTPProxy 1.2.1 as a SIP Proxy. I want to ensure that all my sip and media (RTP) passes through the SIP Proxy (Kamailio + RTP Proxy) while we my SIP Client sends any SIP and RTP to SIP Server (asterisk). So it is like below: SIP Agent <--------SIP+RTP---------->SIP Proxy(Kamailio+RTPProxy)<------SIP+RTP-------->SIP Server(Asterisk). Now it is working fine with my current configuration on Kamailio with RTPProxy with SIP and RTP both if my SIP Agent is on the 802.11 LAN that is my PC have local IP address like 192.168.1.2. Now I am facing the problem in the case where my PC is accessing the internet through the USB data card (or Internet USB dongle). Here my PC IP is like 116.203.51.209. In this case my SIP is successfully passes through the SIP Proxy and works good. But in this case my RTP is does not passes through the SIP Proxy (RTPProxy). The SIP Proxy(Kamailio + RTPPRoxy) and SIP Server is running on the public IP. Please find attached my kamailio.cfg. Please let me know what is the issue. Thanks and Regards Varun

1 0

No audio issue
by jaflong jaflong 08 Apr '14

08 Apr '14

Hi, I am at the point where connection is established and no apparent errors are reported. However audio is not output. The rtp traffic seems to be transfering between the points as conclueded because Asterisk debug log shows Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868, ts 221760, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001383, ts 1917269534, len 000160) Sent RTP packet to 10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869, ts 221920, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001384, ts 1917269694, len 000160) Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870, ts 222080, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001385, ts 1917269854, len 000160) And the browser machine on the other endpoint on a tcpdump does shows traffic on the port (41143) What could be causing there to be no audio? This is the connected sdp =0 o=root 350315728 350315728 IN IP4 10.31.xxx.xxx s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.31.xxx.xxx t=0 0 m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2 a=ice-pwd:12611b8146599a9019d59b4b649a7970 a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv

2 2

P-CSCF using MYSQL database
by Andrea Meroni 08 Apr '14

08 Apr '14

Hi all, i am trying to configure an IMS Core Kamailio-based: while configuring the P-CSCF, i set up the parameters 'db_url' and 'db_mode' of the module ims_usrloc_pcscf respectively to 'mysql://kamailio:kamailiorw@localhost/kamailio' and 1 (because i want a DB for persistence). I had created successfully the DB using kamdbctl, and loaded the module 'db_mysql' in kamailio.cfg. Then i tried to run Kamailio, and i got this error: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query: driver error on query: Unknown column 'aor' in 'field list' What am i missing? Consider then when i created the DB i answer yes to all the questions about creating additional tables ... Andrea

2 24

sql_xquery() and xavp checks
by Alex Balashov 08 Apr '14

08 Apr '14

Hi, When using sql_xquery() like this: sql_xquery("ca", "SELECT * FROM gateways", "gateways"); ... what's a good way to check if any rows were returned? Since one does not have a $dbr(gateways=>rows) value in this scenario, what should one do? - is_avp_set("$xavp(gateways=>id)")) does not appear to operate on XAVPs, or at least, the fixup functions reject them: ERROR: avpops [avpops.c:935]: fixup_is_avp_set(): bad attribute name <$xavp(gateways=>id)> - the 'defined' operator does not appear to return a negative condition here: if(!defined $xavp(gateways=>id)) This condition evaluates to true. Much appreciated! -- 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/

4 14

kamailio / rtpproxy - remove codec
by Oliver Roth 08 Apr '14

08 Apr '14

Hi all We use kamailio 3.3.7 and rtpproxy for enduser call-termination. In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38. As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488. How is it possible to remove the whole T38 part of this invite? We tried sdp_remove_codecs_by_name(list) without success - what "name" should we use for T38? [T38, t38, t.38, T.38, ...] sdp_remove_line_by_prefix(string) sdp_remove_media(type) None of these functions did really work - best was the last one with type=image but then the sip header is malformed. As we saw with Kamailio version 4.1.x there are a lot of new functions within sdpops. Would an upgrade help? So basically the question is: How to remove the t38 part of the fax invite? (see attachment) KR, Oli [cid:image001.png@01CF523C.988433F0]

2 3

SIP 480 - Service unvailable
by Amogh meshram 07 Apr '14

07 Apr '14

Hello everyone, >From today i started getting SIP 480 error when making outbound call. I checked out with a different voip provider and got SIP 500 error. Internet calling is working fine. The service (outbound calls) was working fine till yesterday. I'm a newbie to kamailio and voip, so need your help. Please guide me to resolve this error. Thanks in advance.

2 11

SIP client TCP behaviour
by Alex Balashov 07 Apr '14

07 Apr '14

Hello, I don't know a lot about SIP with TCP, so I thought I'd ask for some opinions here. I've been testing Kamailio with TCP against the Android SIP client (4.4.2) and have found, to my annoyance, that it sends a RURI with a transport attribute in its INVITEs: INVITE sip:14045551212@sip.evaristesys.com;transport=tcp My upstream gateways are all UDP, so this causes Kamailio to attempt to contact them all via TCP (and fail). I know how to bend the transport with Kamailio, and I can strip this attribute. My question is more methodological: is this "correct" for a client to do? In my opinion, the answer is no. The client should not be so presumptuous. It should specify the transport in its Contact, to indicate how it wants to be reached, and leave other decisions about transport to the UAS. But is there something I am perhaps missing? Thanks, -- 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 4

TLS connection with short lifetime - how to handle it?
by Dmytro Bogovych 07 Apr '14

07 Apr '14

Greetings. I have the next problem: iOS based clients connect via TLS to kamailio server. They run mostly in background mode - it means connection refresh interval is ~10 minutes. Some of clients reside behind paranoidal routers which considers such idle connections as lost and closes them. I see only way to resolve it is to send OPTIONS/NOTIFY/CRLF from kamailio to clients. Tried to use nat_traversal module by calling nat_keepalive() to every REGISTER message. But there is no incoming keepalive (OPTIONS or NOTIFY) on client side. Is there another way to make "heartbeat" in SIP/TLS from server side? Thank you.

1 0

One-way Relay through RTPproxy
by Nandini madhu 07 Apr '14

07 Apr '14

Dear All, I am running Kamailio (4.0) and RTPproxy server (1.2.1) server in my set-up. When i call from Client A to B , the RTP packets are proxying trhrough RTPproxy service to other end client, but no flow from client B to client A. With this calls Are unsuccessful. I traced a SIP capture (please find in the attachment) and analysed it, as in there only one client 's (A) IP address is changed to RTPproxy IP address in SDP body. What can be do to resolve this issue ? Please anybody help me And also find the attachment for my kamailio config file, and suggest me or point me about any misplacement of NAT traversal function call in the script. Please consider this and help me out as i am newbie in scripting. Regards, Nandini

2 2

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