- sr-users - kamailio.org

Configuring/Integrating Kamailio Open IMS with FHoSS
by Olvera, Victor 04 Nov '16

04 Nov '16

How does one configure Kamailio Open IMS and FHoSS? In other words, how does one configure IMS and FHoSS so that when soft calls are made they must be processed by the FHoSS before they go through the Open IMS server? Great regards, Victor Olvera

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nathelper not changes IP
by Yuriy Gorlichenko 03 Nov '16

03 Nov '16

I trying to use fix_nated_sdp with 0x02 and 0x08 flags for changing ip addres and the SDP body part like fix_nated_sdp(10,"1.2.3.4") But for now it beaks SDP At the output i see next o=- 7300689428214760503 2 IN IP4 1.1.1.1 c=IN IP4 1.1.1.1 1.2.3.41.2.3.4 <- is just a line that added after fix_nated_sdp atthe end of SDP body kamailio -v version: kamailio 4.4.3 (x86_64/linux) e91aec flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e91aec compiled on 07:26:14 Oct 18 2016 with gcc 4.9.2

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Cannot send request to OpenSIPS
by Laura 02 Nov '16

02 Nov '16

Dear support, on my Kamailio server (version: kamailio 4.3.6 (x86_64/linux) ) i see lot of these messages.. Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: cannot write request to `/var/run/opensips/socket': No such file or directory Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: failed to end dialog: Cannot send request to OpenSIPS I have only configured Disptachet (mediaproxy) for write on db.. Any idea ? Thanks Laura

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diff between location and kamctl ul show
by Никитенко Виталий 02 Nov '16

02 Nov '16

Hi all! I have kamailio 4.2.3 with options modparam ( "usrloc", "db_url", DBURL) modparam ( "usrloc", "db_mode", 2) But not all registred users displayed in table "location". When viewed "kamctl ul show" displays Domain :: location table = 1024 records = 1172 max_slot = 5 But "location" table has 651 rows. Why not all users saved to mysql table? Thank!

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Kamailio TLS client config selection
by Sergey Basov 02 Nov '16

02 Nov '16

Hi Daniel. After some try to configure kamailio 4.4.3 to act as SPI TLS client for Cisco SIP TLS gateways I have found one issue. If I do client configuration for tls [client:10.1.23.19:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca1.pem [client:10.1.23.29:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca2.pem [client:default] verify_certificate = no require_certificate = no Kamailo always do default profile selection (I do configuration without server_name or server_id, with it kamailio works fine but there are some troubles to make selection of this parameters from config script, I need additional checks and queries) after some research in tls module source code I have added some debug information in file tls_server.c: if (c->flags & F_CONN_PASSIVE) { state=S_TLS_ACCEPTING; dom = tls_lookup_cfg(cfg, TLS_DOMAIN_SRV, &c->rcv.dst_ip, c->rcv.dst_port, 0, 0); } else { state=S_TLS_CONNECTING; sname = tls_get_connect_server_name(); srvid = tls_get_connect_server_id(); // ------------------------------------------------------------- DBG("Entered client config loockup (c->rcv.dst_port %d)\n", c->rcv.dst_port); DBG("Entered client config loockup (&c->rcv.dst_ip %s)\n", ip_addr2a(&c->rcv.dst_ip)); DBG("Entered client config loockup (c->rcv.src_port %d)\n", c->rcv.src_port); DBG("Entered client config loockup (&c->rcv.src_ip %s)\n", ip_addr2a(&c->rcv.src_ip)); // ------------------------------------------------------------- dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); } After making Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <INVITE> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:9098@10.1.23.19:5061;transport=TLS> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> I see Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 10.1.23.19 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:985]: tcpconn_new(): on port 5061, type 3 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:1295]: tcpconn_add(): hashes: 1394:0:0, 1 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:160]: tls_get_connect_server_name(): xavp with outbound server name not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:140]: tls_get_connect_server_id(): xavp with outbound server id not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:219]: tls_complete_init(): Entered client config loockup (c->rcv.dst_port 40123) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:220]: tls_complete_init(): Entered client config loockup (&c->rcv.dst_ip 10.1.23.23) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:221]: tls_complete_init(): Entered client config loockup (c->rcv.src_port 5061) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:222]: tls_complete_init(): Entered client config loockup (&c->rcv.src_ip 10.1.23.19) Where: &c->rcv.dst_ip 10.1.23.23 - it is my local kamailio tls socket ip address to make tls connect from c->rcv.dst_port 40123 - it is my local kamailio tls socket port &c->rcv.src_ip 10.1.23.19 - ip of my TLS device to make tls connection to c->rcv.src_port 5061 - port of my TLS device to make tls connection to so if I change line dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); to dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.src_ip, c->rcv.src_port, sname, srvid); I got correct client domain selection Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:233]: tls_complete_init(): Using initial TLS domain TLSc<10.1.23.19:5061> (dom 0x7fd2eefa3d68 ctx 0x7fd2ef7e70a8 sn []) Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started Can you look at this code? Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com tel: (+38067) 403-62-54

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Kamailio + IMS + sipml5 call to softphone
by Serhat Guler 01 Nov '16

01 Nov '16

Hi, I have a setup as follows: IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for webrtc calls. Calls(both audio and video) between to sipml5 clients using firefox web browser is possible. The session is setup for the calls from sipml5 to Mercuro, but then there isn't audio flow as the codecs are not compatible. Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS codecs as firefox but this time the session isn't being setup. Boghe replies with "Reason: SIP; cause=488; text="Bad content" " I have seen a similar issue has been mentioned here: https://github.com/c00lz3r0/boghe/issues/157 but the initial invite request from sipml5 does have the SDP with media attributes.

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ACC module and BYE attacks
by Gholamreza Sabery 01 Nov '16

01 Nov '16

If you enable acc module to log CDR data into a database a user can send an infinite number of BYE requests to the server and all of them will be logged into the database as "481 Call/transaction does not exist"! How can we prevent this?

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/usr/sbin/kamailio - unresolvable destination error
by Ginti Saurabh 01 Nov '16

01 Nov '16

Hi, Please help to sort the problem. Getting "Unresolvable destination" error in /var/log/messages.

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Terminating PCSCF unresolvable request URI (Kamailio IMS Websockets)
by Serhat Guler 01 Nov '16

01 Nov '16

Dear all, I am trying to make a call between 2 webrtc clients(sipml5) using my kamailio IMS setup. The invite request fails at the terminating PCSCF with error code "Status: 478 Unresolvable destination (478/TM)". I have traced the packages and saw that the package that arrives at the terminating PCSCF has an unresolvable request URI due to the websocket connection. The package being sent from terminating ICSCF to SCSCF has the "Request-URI: sip:bob@net1.test" The package from terminating SCSCF to terminating PCSCF has the "Request-URI: sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws" I guess the terminating SCSCF replaces the request URI with the one it stored during the registration process, but the terminating PCSCF cannot resolve the host name&port number; therefore, it reports back "Unresolvable destination". Is that really the case happening in here ? How can I get over this problem ? Any ideas are welcome. Cheers, Serhat

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Server error for SIP client
by Hermann Norpois 01 Nov '16

01 Nov '16

Hello, if my SIP client tries to connect ... a server error occured (see below). Could you please help me to figure out the reason? Kamailio is installed on an amazon instance with an elastic IP (52.208.249.214, hostname: synth.ddns.net). Thanks Hermann ubuntu@ip-172-31-32-10:~sudo /etc/init.d/kamailio restart * Stopping Kamailio SIP Server: kamailio [ OK ] * Starting Kamailio SIP Server: kamailio loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 172.31.32.10:5060 tcp: 127.0.0.1:5060 tcp: 172.31.32.10:5060 Aliases: tcp: ip-172-31-32-10.eu-west-1.compute.internal:5060 tcp: localhost:5060 udp: ip-172-31-32-10.eu-west-1.compute.internal:5060 udp: localhost:5060 *: synth.ddns.net:* ubuntu@ip-172-31-32-10:~$ kamctl db show subscriber +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | Johanna | synth.ddns.net | 12345 | | 4fa26cedd456d7105c01a48742e71e42 | 1e249ac341d95701f855fce18bec0fba | NULL | ubuntu@ip-172-31-32-10:~$ sudo ngrep -d any -qt -W byline . port 5060 Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. . U 2016/10/20 13:44:33.727161 52.208.249.214:5060 -> 172.31.32.10:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2605d82fd11bd8f788cfad582884eb6d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.e425ad98e0436abc4661d1452a49670d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2d047656fb9df0edfccd4a40d20237b7.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.f54ab9c35b22814b15acbe16bcae387f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.db5960988332c2f4899a5e1c2b57e3c6.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.62f1191bd5af8480698ce118984536c0.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. 2) My logfile /var/log/kamailio/kamailio.log is empty though /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log And in /etc/kamailio/kamailio.conf: log_facility=LOG_LOCAL0 Why is the logfile empty? Thanks Herman

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Store and add custom variables for sip registrations
by Markus Bönke 01 Nov '16

01 Nov '16

Hallo all, we want to use kamailio as registrar server in front of freeswitch. When a user registers we need to lookup and store some user variables. To every INVITE that is received from a registered user we want to add the variables as x sip headers before we relay them to freeswitch. The question is what is the best way to implement this? We tried to store the variables in a htable on register and then lookup and attach the variables on every INVITE from there. Is this the correct approach, one issue we see is that htable is not writing new inserts directly to the database. I’ve also seen that there is a table location_atts in the kamailio db which looks like that it is intended to store custom key value pairs for registrations, but the seems to be no module which writes to that table, or is there one? Thanks for any help or suggestions Markus

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TSILO module use failure
by Igor Olhovskiy 31 Oct '16

31 Oct '16

Hi! I’m trying to implement scheme much like Federico Cabiddu is described here https://www.youtube.com/watch?v=4XIrR9bwUkM Done a bit modification to his script, cause it’s not fully correct on 4.4 version So, parts of config looks like …. route[REGISTRAR] { … if (!save("location")) { sl_reply_error(); } else { route(PUSHJOIN); } exit; …. } route[LOCATION] { if (!lookup("location")) { …… if (is_method("INVITE")) { send_reply("100", "Trying"); route(SUSPEND); } } else { if (is_method("INVITE")) { if (t_newtran()) { ts_store("sip:$tU@$fd"); } $sht(vtp=>stored::$rU) = 1; } } route(RELAY); } ….. route[SUSPEND] { if(!t_suspend()) { send_reply("501", "Unknown destination"); exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); } # append branches or resume the transaction route[PUSHJOIN] { … #if was suspended - recover, if not - ts_append ….. ts_append(«location", "sip:$tU@$fd"); return; … t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } # lookup and relay after resuming transaction route[INVRESUME] { lookup("location"); t_relay(); ts_store("sip:$tU@$fd"); $sht(vtp=>stored::$rU) = 1; } Branches on ts_store are saved in format (sip:username@domain) and also in this format they are added on ts_append. All regarding to routes working ok, except on ts_append I got ERROR: tm [t_append_branches.c:168]: t_append_branches(): ERROR: t_append_branch: failure to add branches (-1) Also, it’s a mixed environment, like wss + udp + tcp. Any idea why? Thanks. -- Best regards, Igor

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Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 30 Oct '16

30 Oct '16

Hi all, I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great. Thanks, Serhat On 24 October 2016 at 21:00, Serhat Guler <srtguler(a)gmail.com> wrote: > Hi Daniel, > > I am using only record_route() without any parameters. I do not have a > proper computer atm to draw the network diagram, but I can tell you shortly > about the network setup. > > I have only enabled websockets for the pcscf to allow ws and wss > connections. In that case there is a ws connection that uses UDP protocol. > This is the ACK to complete the session setup. > > the sipml5 client is configured as follows: > WebSocket Server URL: ws://192.168.0.11:880 > SIP outbound Proxy URL: udp://192.168.0.11:4060 > > Mercuro IMS client: uses UDP port as well: 4060 > > The call is made from sipml5 client. The Mercuro phone rings, and when I > reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from > sipml5 doesn't pass the PCSCF as I explained in the previous message. > > A part of PCSCF cfg file: > > # Check for Subsequent requests: > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if ($route_uri =~ "sip:mo@.*") { > setflag(FLT_MO); > } > if(!isdsturiset()) { > handle_ruri_alias(); > } > # RTP-Relay, if necessary > route(RTPPROXY); > t_relay(); > } else { > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > xlog("L_INFO", "ACK without matching transaction ... > ignore and discard!!!!!\n"); > # ACK without matching transaction ... ignore and > discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > > Cheers, > Serhat > > > > On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I haven't noticed the log files, it's ok. >> >> From the Route header, I see that there is a proxy that uses WS: >> >> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >> 1nCMEI1mR0RztrB;did=e82.0c3> >> That is the address of the next hop and typically a proxy doesn't use >> websocket connection to another proxy. Can you show a diagram with the sip >> server nodes in your network and what protocols are used between them? >> >> Are you simply use record_route() function, or some other function or >> different parameters to it? >> >> Cheers, >> Daniel >> >> >> On 24/10/16 12:18, Serhat Guler wrote: >> >> Hi Daniel, >> >> Thanks for your reply. I actually attached a log file with debug level 3, >> consisting ACK related messages. If you would like to see more logs, I'll >> send a new log file in the evening. >> >> Cheers, >> Serhat >> >> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> can you get all the log messages for ACK but with debug=3 in the >>> kamailio.cfg? >>> >>> Cheers, >>> Daniel >>> >>> On 23/10/16 22:04, Serhat Guler wrote: >>> >>> Hello, >>> >>> I finally managed to place a call from sipml5 webrtc client to Mercuro >>> IMS client. The phone rings, and when I answer it sends 200 OK to the >>> sipml5 where as sipml5 send back an ACK message which never passes the >>> originating PCSCF. The PCSCF says: >>> >>> 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS >>> connection (id: 0) for WebSocket could not be found >>> 8(3640) ERROR: <core> [msg_translator.c:1947]: >>> build_req_buf_from_sip_req(): could not create Via header >>> 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building >>> failed >>> >>> I doubt that the WebSocket connection is closed, cause when I terminate >>> the call from Mercuro client a bye request is being sent to the sipml5. >>> >>> The ACK package: >>> >>> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. >>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 >>> hG4bKvuly7bmxnN4aqM4zZTIS;rport >>> From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB >>> To: <sip:alice@net1.test>;tag=18823 >>> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c >>> all=no;transport=ws>;+g.oma.sip-im;language="en,fr" >>> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 >>> CSeq: 3887 ACK >>> Content-Length: >>> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> >>> Max-Forwards: 69 >>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >>> 1nCMEI1mR0RztrB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz >>> trB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.1c3> >>> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 >>> Organization: Doubango Telecom >>> >>> Have been thinking for quite a while, but couldn't really find a reason >>> why it wouldn't add the v,a header. A debug 3 level log file is also >>> attached. >>> >>> Thanks in advance, >>> Serhat >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >>> >>> _______________________________________________ SIP Express Router >>> (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cg >>> i-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >> >> >

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digest authentication based on two password columns
by Jayesh Nambiar 28 Oct '16

28 Oct '16

Hello, I was wondering if it is possible to do a failover in digest authentication. For eg: I get two possible valid password values. Can I do a pv_proxy_authenticate in such a way where if authentication fails, I do a pv_proxy_autenticate again on the different value to check if that succeeds? Is it even allowed in digest authentication mechanism. Just getting opinions. Thanks for any help. - Jayesh

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Re: [SR-Users] fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Daniel, No that exactly the one, Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 28 October, 2016 07:37:31 Subject: Re: [SR-Users] fetch contact Hello, the contact address is one for attributes in $ulc(...), iirc, it is the =>addr -- see the wiki for pv cookbook for more details. Or are you looking for something else? Cheers, Daniel On 27/10/16 21:43, Slava Bendersky wrote: Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks _______________________________________________ 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Errors when parsing Diversion header with multiple comma-separated diversions
by Björn Bylander 28 Oct '16

28 Oct '16

Hello! Given a Diversion header like the following: Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Kamailio 4.4.3 (and probably earlier versions) emit error messages complaining about the comma character. According to RFC 6044 multiple comma-separated diversions in a single Diversion-header is allowed. Full INVITE as an example: INVITE sip:+1234567890@127.0.0.100:5060;user=phone;transport=tcp SIP/2.0 Via:SIP/2.0/TCP 127.0.0.101;branch=z9hG4bKBroadWorks.-1od4eg8-127.0.0.100V5060-0-82844591-13 54681939-1477634439005- From:<sip:+9876543210@127.0.0.101;user=phone>;tag=1354681939-1477634439005- To:<sip:+1234567890@127.0.0.100:5060;user=phone> Call-ID:BW0800390052810161562691307@127.0.0.101 CSeq:82844591 INVITE Contact:<sip:127.0.0.101:5060;transport=tcp> P-Asserted-Identity:<sip:+9876543210@127.0.0.101;user=phone> Privacy:none Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE X-BroadWorks-DNC:network-address="sip:+9876543210@127.0.0.101;user=phone" Accept:application/dtmf-relay,application/media_control+xml,application/sdp, application/x-broadworks-call-center+xml,multipart/mixed Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:267 v=0 o=BroadWorks 100442303 1 IN IP4 127.0.0.101 s=- c=IN IP4 127.0.0.102 t=0 0 m=audio 29778 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 Example log messages: Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:488]: parse_to_param(): ERROR parse_to_param : invalid character ',' in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered>> Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:254]: parse_to_param(): ERROR: parse_to_param :unexpected char ["] in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered,>> . Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:906]: parse_addr_spec(): ERROR: parse_to: invalid To - unexpected end of header in state 21 Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_diversion.c:69]: parse_diversion_header(): ERROR:parse_diversion_header: bad diversion header Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (562,21-562,35) The code on line 562 is $avp(div_user) = $(di{uri.user});. Can this be handled in any way by Kamailio without changing the parser? Best regards, Björn Bylander

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fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks

2 1

logfile is empty
by Hermann Norpois 27 Oct '16

27 Oct '16

my logfile /var/log/kamailio/kamailio.log is empty. 1) /etc/rsyslog.d/kamailio.cfg exists with the entry: log_facility=LOG_LOCAL0 2) /etc/rsyslog.con has the entry: # # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log # local0.* -/var/log/kamailio/kamailio/kamailio.log Alternatively I edited /etc/rsyslog.d/kamailio.conf that should be used by rsylog.conf 3) /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log I am working with ubuntu on an amazon instance. Why is the logfile empty? Thanks Herman

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How do I see the full SDP?
by Martin Taschen 27 Oct '16

27 Oct '16

Hello, kamailio seems to work fine. After registration I would like to see the complete SDP of an user. . How is this done? Thanks Martin Taschen

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Re: [SR-Users] sr-users Digest, Vol 137, Issue 25
by surendra tiwari 27 Oct '16

27 Oct '16

i used the cnxcc module its successfully terminated the call. but if you want i can help you. On Thu, Oct 27, 2016 at 12:58 AM, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: cnxcc doesn't terminate calls (Igor Potjevlesch) > 2. logfile is empty (Hermann Norpois) > 3. Re: Out of memory in UB 210: OOM killed process 12261 > (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB > (Jurijs Ivolga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2016 12:48:29 +0200 > From: "Igor Potjevlesch" <igor.potjevlesch(a)gmail.com> > To: "'Kamailio $SER$ - Users Mailing List'" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] cnxcc doesn't terminate calls > Message-ID: <009c01d22f76$7d0c9a30$7725ce90$(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > > > Yep, the dialog is created. I put the dialog into DB and see it. > > Nothing appears in the logs when the timer is reached. > > > > I succeed to do the same behaviour with dialog module only and > dlg_set_timeout. Looks to do why I want. > > > > I'm a bit frustrating about cnxcc because I really don't understand why > nothing happen after max_time, but, anyway, it works with dialog only. > > > > Regards, > > > > Igor. > > > > De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part > de Arsen > Envoyé : mercredi 26 octobre 2016 11:15 > À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> > Objet : Re: [SR-Users] cnxcc doesn't terminate calls > > > > Hi Igor, > > > > Make sure that you create a dialog for this call and $dlg_var(key) is > available (This pseudo-variable will be available only for subsequential > requests after doing loose_route().) > > > > Also check kamailio logs maybe you try to terminate call for wrong > customer id. > > > > On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch < > igor.potjevlesch(a)gmail.com <mailto:igor.potjevlesch@gmail.com> > wrote: > > Hi, > > > > I'm testing cnxcc module in order to make a simple call duration > limitation. > > My understanding is to use the function like this : > > > > $dlg_var(customer_id) = "customer-" + $fU; > > $var(max_time) = 5; > > > > if (!cnxcc_set_max_time("$dlg_var(customer_id)", > "$var(max_time)")) > > > > Unfortunately, after 5 seconds, nothing happen. I don't even see an error > from dialog module. > > > > Anyone has a better understanding of the way the module runs? Thank you! > > > > Regards, > > > > Igor. > > > _______________________________________________ > 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 > > > > > > > > -- > > Regards, > Arsen. > >

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cnxcc doesn't terminate calls
by Igor Potjevlesch 26 Oct '16

26 Oct '16

Hi, I'm testing cnxcc module in order to make a simple call duration limitation. My understanding is to use the function like this : $dlg_var(customer_id) = "customer-" + $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$dlg_var(customer_id)", "$var(max_time)")) Unfortunately, after 5 seconds, nothing happen. I don't even see an error from dialog module. Anyone has a better understanding of the way the module runs? Thank you! Regards, Igor.

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app_perl modifies Via header?
by Igor Potjevlesch 25 Oct '16

25 Oct '16

Hello, I try to use app_perl and cnxcc module. As soon as I load the module like this: # --------- app_perl params ---- modparam("app_perl", "filename", "/usr/local/etc/kamailio/AAA/aaa-script.pl") modparam("app_perl", "modpath", "/usr/local/lib64/kamailio/perl/") The Via header is modified for each calls: /usr/local/sbin/kamailio[21064]: INFO: <core> [parser/msg_parser.c:355]: parse_headers(): ERROR: bad header field [Via:ASIP/2.0/UDP 91.] /usr/local/sbin/kamailio[21064]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <A> on state 100 What could explain this behaviour? Regards, Igor.

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Kamailio SIP TLS issue
by Sergey Basov 25 Oct '16

25 Oct '16

Hi All I have some strange behavior of kamailio with TLS. I have configured second IP addres on server, added it to tls listener, and tls.cfg file. But when I try to connect using openssl s_client -showcerts -connect 10.1.23.33:5061 -tls1 -state and openssl s_client -showcerts -connect 10.1.23.23:5061 -tls1 -state I see same certificates (sip2 my config samples are bellow) if I make changes in port number (for ip 10.1.23.33 set port 5091 in both config parts) - I see correct certificates. Does anyone have this problem? Thanks in advance. ----- listen section ---- listen=tls:10.1.23.23:5061 listen=tls:10.1.23.33:5061 ----- tls.cfg ------ [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.33:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.23:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip2.key certificate = /etc/kamailio/keys/sip2.crt -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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Sending INVITE’s on REGISTER
by Igor Olhovskiy 25 Oct '16

25 Oct '16

Hi! I have this kind of scenario Receive an INVITE -> send it to user, but if possible, if I’ll receive REGISTER with same credentials I’m calling to - drop an INVITE there also, if call was not answered on first location. Means create other INVITE branch async on the fly, if possible. Where at least to look at? Thanks. -- Best regards, Igor

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How compile Kamailio in freebsd : add carrierroute module
by AFRICALLSHOP 25 Oct '16

25 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I do : 1) cd /usr/ports/net/kamailio 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

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How compile Kamailio in freebsd : add carrierroute module
by Abdoul Osséni 24 Oct '16

24 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I did : 1) cd /usr/ports/net/kamailio/ 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

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Rewriting To on BYE
by Charles Phillips 24 Oct '16

24 Oct '16

I need a sanity check if you don’t mind… I am setting up a stateless proxy for several (separate, not load balanced) Asterisk servers behind Kamailio 4.2 (with domain and dispatcher). All works as expected except when called party hangs up, as Asterisk seems to address the BYE “To” (that is sent to the calling party) to the originating domain, and not the IP of the UA (as it does in other cases). When this happens, stateless Kamailio does not seem to know how to route them, which makes sense. I assume I would need to rewrite the “To” field on BYE message so that it contains the UA’s IP (to match the BYE’s that are routed correctly). The RURI header contains the IP/port of the UA, so I am attempting to use it and here is where I am stuck… Here is the subst from onreply_route: if (is_method("BYE")) { xlog("L_NOTICE", "BYE Detected \n"); subst('/^To:(.*)<sip:(.*)@(.*)>(.*)$/To: <sip:\2@$rd:$rp>\4/ig'); } Here is the “To”: To: "Test User 2" <sip:102@zxcv.asdf.qwerty.net>;tag=8AB53900B9646F1DF05330E7FBF846C8 Any obvious glaring mistakes here? Would I then need additional logic to forward the BYE on? My apologies if this is wordy, or nonsensical...

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PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 24 Oct '16

24 Oct '16

Hello, I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says: 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5. The ACK package: ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB To: <sip:alice@net1.test>;tag=18823 Contact: "Bob"<sip:bob@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> Max-Forwards: 69 Route: <sip:mo@192.168.0.11:880 ;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:4060 ;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:4060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached. Thanks in advance, Serhat

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configuration kamailio 4.4.2 with Asterisk 13.10.0
by Serge Gabriel Ahnoux 24 Oct '16

24 Oct '16

Hello I am trying to configure kamailio 4.4.2 with asterisk 13.10.0. My Os is Debian 7.4 I followed the tutorial : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… The two softwares are installed on my server. So they have the same IP adress but the port is different like they said in the tutorial. I use too Mysql to stock users informations I have 2 databases Kamailio and Asterisk. I use ODBC for communicate beetwen asterisk server and his database. However when my sip client (linephone) connects to kamalio, kamailio doesn't forward information to asterisk. I don't kow the reason. So i can send message but not call If someone knows how to configure, please help me. I join my kamailio configuration file. Regards. ahnouxs(a)gmail.com

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Dropping non-provisional replies in onreply_route
by Alex Balashov 21 Oct '16

21 Oct '16

The core documentation says that in a named onreply_route[], only provisional replies can be drop()'d. To drop any reply, it is necessary to use a global onreply_route. Is there any workaround for this, i.e. so I can drop a 2xx reply from a specific TM transaction? The reason is, to know whether to drop it, I need access to either AVPs or, ideally, dialog variables. Since the global onreply_route is executed by the core, I presume I won't have access to anything that persists through TM there. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Edge proxy / Registrar Script
by Infinicalls Infinicalls 21 Oct '16

21 Oct '16

Hi, I am trying to get a simple Edge Proxy - Registrar setup working behind NAT. Initially I tried various scripts - including the one mentioned in the Oubound page and Peter Dunkley' example - but got some errors related to PATH. Can somebody give me a working script of both proxy and registrar? Any help would be gladly appreciated. regards Ganesh Kumar -- --- http://www.infinicalls.com

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Basic TM suspend question
by Alex Balashov 21 Oct '16

21 Oct '16

Hi, If a transaction is currently t_suspended() and has not been t_continued(), will t_check_trans() still find that transaction for retransmission-dampening purposes, e.g. if a retransmission of the same request as the original one is received? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Postgres: Exclusive Transaction Locks vs. Vacuumer
by Stefan Kohlhauser 21 Oct '16

21 Oct '16

Hey everyone! We noticed that under high load of registrations and using the pua module we can reach a state where the PUA table bloats our Postgres to several GB. However, a SELECT on pua only shows a few hundred entries. The other half million or so entries are obsoleted but not cleaned up. The reason for this seems to be that the autovacuumer of the Postgres can't get to work because the PUA table is almost permanently locked. Our Postgres logs are full with errors about the autovacuumer being canceled because it couldn't get to work. Looking at modules/db_postgres/km_dbase.c::db_postgres_start_transaction() I noticed that there are two locking modes used: str lock_write_end_str = str_init(" IN EXCLUSIVE MODE"); str lock_full_end_str = str_init(" IN ACCESS EXCLUSIVE MODE"); According to https://www.postgresql.org/docs/9.2/static/explicit-locking.html these modes mean forbidding everything else except reads and forbidding everything else. However, according to this documentation ROW EXCLUSIVE is the standard lock for entry altering operations (INSERT, UPDATE, DELETE). And this would still allow the autovacuumer to do its job. Is there a specific reason why these more restricting locking modes are used instead of ROW EXCLUSIVE? Thanks. Best regards, Stefan

2 1

TCP FIN after 10 minutes
by gmele 20 Oct '16

20 Oct '16

Hello, we have deployed a Kamailio acting as SIP proxy on a RHEL 7.2 machine. Clients (mainly mobile phones) connects to the proxy using a TLS protected TCP connection. In the kamailio config, we've set : #!ifdef WITH_TLS enable_tls=yes tcp_async=yes tcp_connection_lifetime=3605 tcp_accept_no_cl=yes tcp_crlf_ping = yes #!endif Our problem is that, even if we set the tcp_connection_lifetime to > 1 hour, the tcp connection is closed after 10 minutes: on tcp dumps, we see clearly the TCP FIN sent by machine hosting the kamailio proxy... Setting parameters tcp_keepidle/keepintlv/keepcnt in the kamailio config didn't change the behavior. We also set TCP keepalived at system level, but without result... This TCP closure is causing us lot of problems when calls between 2 UAs last more than 10 minutes because the REINVITE or BYE messages are lost. Also, closing the TCP connection will wake up the mobile app and make it resend a REGISTER, thing we absolutely want to avoid. Is there a parameter we can use to avoid this closure? I had a look in previous posts, found people with the same problem as mine, but no real solution... Thx for your help. Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TCP-FIN-after-10-minutes-tp152646.h… Sent from the Users mailing list archive at Nabble.com.

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Re: [SR-Users] help with NOTIFY
by Slava Bendersky 20 Oct '16

20 Oct '16

Hello Everyone, This message as continue conversation from http://lists.sip-router.org/pipermail/sr-users/2015-March/087557.html. That my previous post about it. I never was be be able forward NOTIFY from asterisk to client through kamailio. Right now in use asterisk 14 pjsip. Any help thank you. volga629 From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 19 October, 2016 18:36:52 Subject: [SR-Users] help with NOTIFY Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

sipml5 fail invite kamailio ims web sockets
by Serhat Guler 20 Oct '16

20 Oct '16

Dear all, I can register the sipml5 client to my kamailio IMS setup successfully, but when I try to make a call the invite fails with "478 Unresolvable destination". From the wireshark trace file I think the PCSCF on the terminating side cannot resolve the destination. After the invite request fails the caller re-registers to the networkj automatically. The wireshark file is attached. The registration process happens successfully but all of the nodes give some errors: 10(3773) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=425 a=25 n=xlog 10(3773) ERROR: <script>: REGISTER (sip:alice@net1.test (192.168.0.10:14115) to sip:alice@net1.test, 25c80750-d52d-30c8-d373-c0136401a927) 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=5 n=route 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=622 a=25 n=mf_process_maxfwd_header 10(3773) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <59840> <REGISTER> 10(3773) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 10(3773) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=26 n=sanity_check 10(3773) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 10(3773) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=24 n=has_totag 10(3773) DEBUG: siputils [checks.c:97]: has_totag(): no totag 10(3773) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0xb6e5fe40 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=24 n=t_lookup_request 10(3773) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffff 10(3773) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=21728, isACK=0 10(3773) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 10(3773) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 10(3773) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=652 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=650 a=26 n=send_reply 10(3773) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 10(3773) DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [2] 10(3773) DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0xb2329228] refcnt [2] 10(3773) DEBUG: websocket [ws_frame.c:751]: ws_frame_transmit(): Tx message: The SCSCF says (while inviting): 0(3582) ERROR: dialog_ng [dlg_handlers.c:196]: populate_leg_info(): bad sip message or missing Contact hdr 0(3582) ERROR: dialog_ng [dlg_handlers.c:453]: dlg_onreply(): could not add further info to the dlg out Making calls between 2 IMS clients(boghe) is possible. Would appreciate any help (I can share the cfg files if needed). Cheers, Serhat

2 3

Kamailio script config - Generate Documentation
by José Seabra 20 Oct '16

20 Oct '16

Hello there, I'm sending this email in order to know if anyone had work or has any idea how we can generate Documentation from kamailio script using Comments? This would be great because we could write comments explaining the Route method during script development and use it to generate documentation about our kamailio script, saving time and leave our deployment documented. Thank you -- Best Regards José Seabra

3 5

Regarding pua_reginfo, server_address
by Ram Anji 20 Oct '16

20 Oct '16

Hi All, I am configured kamailio to handle reg info event subscriptions, My server is running if i configure the following parameter. #kamailio local ip is 192.168.20.20 *modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.20.20 <sip%3Areginfo(a)192.168.20.20>")* If i am not configured the above server address, i am getting bad configuration Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: auth [auth_mod.c:333]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.allow Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:607]: mod_init(): default allow file (/etc/kamailio/permissions.al... rule set Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.deny Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:616]: mod_init(): default deny file (/etc/kamailio/permissions.den... rule set *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: pua_reginfo [pua_reginfo.c:105]: mod_init(): server_address parameter not set* *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: <core> [sr_module.c:944]: init_mod(): Error while initializing module pua_reginfo (/usr/lib/...ginfo.so)* Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Oct 18 11:31:36 arm kamailio[2771]: already running ... failed! Oct 18 11:31:36 arm kamailio[2771]: . Oct 18 11:31:36 arm systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Hint: Some lines were ellipsized, use -l to show in full. I want to know what is the significance of this parameter and find no where this value is used. Thanks in advance. Regards, Raman

2 1

RTPProxy
by Rodrigo Moreira 20 Oct '16

20 Oct '16

Hi, What is difference between modules rtpproxy and rtpengine? Best regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

5 12

memory is not freed in system, while kamailio's statistic shows OK
by Vasiliy Ganchev 20 Oct '16

20 Oct '16

Hi folk! Have a production server - used as webrtc to sip proxy - it is built on kamailio v 4.4.2 - Server handles connections from ~4000 incoming SIP trunks (transport=TLS). Kamailio provides auth for this trunks. - Kamailio receives also connections from Webrtc app. Calls from Webrtc routed to one of the 4000 sip trunks. Server is running on Debian GNU/Linux 7 The number of calls on the system is small. Most of the load is generated by registration/re-registrations of SIP trunks The problem is following: - output of kamcmd core.shmmem and kamcmd pkg.stats shows constant using of memory - output of htop shows constantly increasing memory usage, e.g. after one week working: PID USER PRI NI VIRT RES SHR S CPU% MEM% TIME+ Command ..... 6013 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:28.80 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6002 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:30.93 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6012 kamailio 20 0 744M 134M 130M S 0.0 3.5 2:35.00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 5991 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:05.88 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6018 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:31.90 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio ..... Despite running with: PKG_MEMORY=32, Amount of used memory per process is higher. Finishes all with monit killing kamailio, because of memory usage more than allowed studied this tutorial: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory but did not find a way how to dig up the problem's root. Tried also to reproduce the issue in lab environment (having clone of production server). Generated load with incoming registration using PJSIP, but with no success - memory usage is constant. kamailio -v version: kamailio 4.4.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 Any suggestion/idea how to troubleshoot deeper would be highly apprecieted Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/memory-is-not-freed-in-system-while… Sent from the Users mailing list archive at Nabble.com.

2 3

help with NOTIFY
by Slava Bendersky 19 Oct '16

19 Oct '16

Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629

1 0

Active calls limit
by Ivan Dudko 19 Oct '16

19 Oct '16

Hello! I am trying to implement limit of active calls for subscribers. And for each subscriber this limit must be personal. I find example route in presentation of Daniel-Constantin Mierla. And i need to set $xavp(caller=>active_calls) = 1; for each dialog. But i can't understand how to load this number of active calls from some-thing similar of user profile or subscriber table? route[DIALOG] { if (is_method("CANCEL") || (has_totag() && is_method("INVITE|BYE|ACK"))) { dlg_manage(); return; } if (is_method("INVITE") && !has_totag() && !isflagset(WITH_ACTIVE_CALLS_LIMIT)) { if( $xavp(caller[0]=>active_calls) != $null && $xavp(caller[0]=>active_calls) > 0 ) { if(!get_profile_size("caller", "$fU@$fd", "$var(acsize)")) { send_reply("500", "No more active calls"); exit; } if($var(acsize)>=$xavp(caller[0]=>active_calls)) { send_reply("403", "No more active calls"); exit; } set_dlg_profile("caller", "$fU@$fd"); } setflag(WITH_ACTIVE_CALLS_LIMIT); dlg_manage(); } }

2 5

Kamailio Bench
by Rodrigo Moreira 19 Oct '16

19 Oct '16

Hello, i need your help, I'm having problems with the reverse response of an INVITE passing through a rtpproxy (NAT). Does anyone have any idea what might be going on? UE is informed of the result INVITE, but the UAS does not receive the test bench detail. Greetings. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Problems with CDRs
by Diogenes Marcano 19 Oct '16

19 Oct '16

I am using kamailio with ubuntu, but I can't obtain CDRs. In my conf. file related to CDR I have: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") But I don't know where is the problem. Regards Diogenes Marcano +58 414 117 2011

2 1

BYE issue
by Nelson Migliaro 18 Oct '16

18 Oct '16

Hello everybody, I am having issues with one SIP vendor. I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk and Media Gateways. Calls get established and I have two way audio but when the remote party hangs up the call, the BYE arrives to the Kamailio and does not move forward. I think the problem is SIP vendor rewrite the BYE header and change the asterisk IP with the public IP of the kamailio. The IP that appears in the header of the BYE have to be the same that appears in the contact (UAC that send the call, in my case the Asterisk). Vendor should not change that IP. ¿Am I correct? Thank you ----------------------------------------------------------------------------------------------------- INVITE ---------------------------------------------------------------------------------------------------- 2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060 INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0 Record-Route: <sip:PUBLIC-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Record-Route: <sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Via: SIP/2.0/UDP PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1 Via: SIP/2.0/UDP PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060 Max-Forwards: 69 From: "SOURCE-NUMBER" <sip:SOURCE-NUMBER@MY-COMPANY>;tag=as5e87b96c To: <sip:DESTINATION-NUMBER@VENDOR-IP> Contact: <sip:SOURCE-NUMBER@PRIVATE-ASTERISK-IP:5060> Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 102 INVITE User-Agent: UAC Date: Mon, 17 Oct 2016 16:53:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 269 v=0 o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP s=Asterisk PBX c=IN IP4 PUBLIC-KAMAILIO-IP t=0 0 m=audio 23456 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes ----------------------------------------------------------------------------------------------------- BYE ----------------------------------------------------------------------------------------------------- 2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060 BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0 Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0 Via: SIP/2.0/UDP VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206 Max-Forwards: 34 Route: <sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> Route: <sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> To: "SOURCE-NUMBER"<sip:SOURCE-NUMBER@YO>;tag=as5e87b96c From: <sip:DESTINATION-NUMBER@PUBLIC-KAMAILIO-IP>;tag=421ce86-co1547-INS001 Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 154701 BYE User-Agent: VENDOR Content-Length: 0 -----------------------------------------------------------------------------------------------------

3 5

420 Bad Extension error
by Infinicalls Infinicalls 18 Oct '16

18 Oct '16

Hi all, I am testing Edge Proxy setup with the example given here : http://kamailio.org/docs/modules/4.2.x/modules/outbound.html and I get this error : U 2016/10/06 09:18:48.331580 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 420 Bad Extension. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK162a.e9f73fe5e52ded1f30801aa00ee2a771.0; received=52.233.25.164. Via: SIP/2.0/UDP 10.114.70.81:1228;received=113.193.149.49;rport=1228;branch=z9h G4bKPj3ec7c87e09d6452b9e68e60052cb2651. From: "User1" <sip:user1@infinicalls-proxy1.myserver.com>; tag=ac1e36aec8bf445bbec8e43cf28e9f48. To: "User1" <sip:user1@infinicalls-proxy1.myserver.com>;ta g=b27e1a1d33761e85846fc98f5f3a7e58.b172. Call-ID: a270d285de6145c6b8670b909b104e74. CSeq: 20149 REGISTER. Contact: <sip:user1@10.114.70.81:1228;ob>;expires=300. Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. Both my proxy and registrar are behind NAT. rtpproxy is installed and started. I already have modparam("registrar", "use_path", 1) in my registrar. Any idea what is missing? Thanks. regards Ganesh Kumar

2 17

Sip route can be as IMS PCSCF node
by Surender Singh 17 Oct '16

17 Oct '16

Hi, Can we use sip router as IMS PCSCF to connectivity with IMS. Does sip router supports the diameter RX interface to connectivity with PCRF . Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)
by Serhat Guler 17 Oct '16

17 Oct '16

Hi, I am trying to register sipml5 webrtc client to my kamailio IMS setup. I have tried to register the client both with ws and wss, but it seems to be that the sipml5 doesn't calculate the authentication digest right. The authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple wireshark file is attached. .10 being the host, .11 being the kamailio server. >From the output of scscf we can see that the digests do not match. [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA= response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=00000001 cnonce=2d4545cf4c935c8094c8b1da3d4a2976 hbody=d41d8cd98f00b204e9800998ecf8427e ims_auth [authorize.c:872]: authenticate(): UE said: 61cbdbeb47c9880ededfca51c3801800 and we expect 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77 (REGISTER) tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established Thanks a lot, Serhat

2 2

How to use sql transformations?
by Daniel Tryba 17 Oct '16

17 Oct '16

How to use the sql transformation? https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transform… has the following example: xlog("$$rm = $rm = $(rm{s.sql})"); But adding this to the request_route and starting kamailio will fail: ERROR: pv [pv_trans.c:2351]: tr_parse_string(): unknown transformation: sql}/sql/3! ERROR: <core> [pvapi.c:1629]: tr_lookup(): error parsing [{s.sql}] ERROR: <core> [pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "rm" ERROR: <core> [pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(rm{s.sql})] at (4) ERROR: xlog [xlog.c:512]: xdbg_fixup_helper(): wrong format[$$rm = $rm = $(rm{s.sql})] ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:372 line 372 is the above xlog and sqlops.so is loaded (and works). Anybody got a working example of this? Or an other hint to prevent sql injections when using user supplied variables in sql queries?

1 0

503 Service unavailable to 500
by Dhruvin Desai 17 Oct '16

17 Oct '16

Hi 503 Service unavailable received at kamailio SIP server is forwarded as 500 server internal error to the UAC. Which seems fine based on the fact that there is no retry-after header. Would like to know if there is any way to to control the SIP Headers like adding Proprietary SIP Headers to the 500 "Server Interrnal Errror" created by kamalio stack for 503 service unavailable (without retry-after Header) It seems that Kamalio does not retain the proprietary sip headers received in 503 to 500 response forwarded. Any insight into the issue would be helpful -- Warm Regards Dhruvin Desai.

2 1

RTPEngine & Homer
by Grant Bagdasarian 17 Oct '16

17 Oct '16

Hello, I'm trying to get rtpengine to send its stats to Homer, but I'm unsure to which IP I should configure RTPengine to send the stats? We have a Kamailio server configured with the sipcapture module, which works perfectly for SIP traffic, but RTP stats are not visible in Homer. Do I need to point RTPengine to the sip capture server or directly to Homer? Running RTPEngine 4.5.2 and Kamailio 4.4.1. Regards, Grant

1 0

HOMER / HEP support in SEMS?
by Daniel Pocock 14 Oct '16

14 Oct '16

Does anybody know if SEMS has HOMER / HEP support like Kamailio, or if it is under development? It would be useful to extract RTCP stats from SRTP streams, the standalone HOMER captagent can't see inside them when they are encrypted (SRTCP) Regards, Daniel

2 1

Accounting CDR generation
by Ivan Dudko 14 Oct '16

14 Oct '16

Hello! I am setting this in kamailio.cfg #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_user=$rU;dst_ouser=$tU;dst_domain=$rd;sip_code=$rs;sip_reason=$rr;callid=$dlg(callid);dst_ip=$Ri") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") #!endif But in CDRs table ('acc_cdrs') field dst_user=$rU is always empty if call is not completed with remote side answer. But in accounting table ('acc') this field isn't empty. Is this behavior is correct? I think it is not correct.

2 7

Websocket with kamailio IMS
by Serhat Guler 13 Oct '16

13 Oct '16

Dear all, I am trying to configure websockets for kamailio IMS. My basic kamailio(without IMS) setup works properly with websockets and without websockets. The IMS functionality also works, but only with IMS clients and now I want to use webrtc clients and make calls. At the moment I am trying to add the websockets support to the pcscf. I have did some changes to the pcscf cfg file. My problem is not really related to the cfg of pcscf as the component is listening for tcp/tls connections, but I cannot connect through the assigned ports. The output of the pcscf component and the the wireshark pcap file for the Websocket are attached. (I basically tried to browse http://server.ip.address:tcp.port or https://server.ip:tls.port without using a webrtc client yet). 192.168.0.10 -> host 192.168.0.11 -> virtual machine running kamailio Thanks in advance, Serhat

2 1

Reload PIPILIMIT table from DB
by Grant Bagdasarian 12 Oct '16

12 Oct '16

Hello, Is it possible to live reload the pipelimit module so it gets new values from the DB without a restart of Kamailio? I couldn't find a command to do this. Regards, Grant Bagdasarian CM

2 3

prefix_route and db_cluster not working
by Martin Nyström 12 Oct '16

12 Oct '16

Hi, I have an issue when I am unable to get prefix_route to work together with db_cluster. If I set the db_url manually it works great, pointing it to a specific database. However we need the redundancy for this module. Dispatcher uses the same cluster without any issues. The following are my configuration sets: # DB CLUSTER (db_cluster.inc) loadmodule "db_cluster.so" modparam("db_cluster", "connection", "KAMDB01=>mysql://kamailio:password123@192.168.55.10/kamailio") modparam("db_cluster", "connection", "KAMDB02=>mysql://kamailio:password123@192.168.55.11/kamailio") modparam("db_cluster", "cluster", "AccCluster=>KAMDB02=9s9s;KAMDB01=4s4s") modparam("db_cluster", "cluster", "Settings=>KAMDB02=9s9s;KAMDB01=4s4s") # KAMAILIO.CFG (kamailio.cfg) # PREFIX ROUTE MODPARAM loadmodule "prefix_route.so" include_file "db_cluster.inc" modparam("prefix_route", "db_url", "cluster://Settings") modparam("prefix_route", "db_table", "prefix_route") modparam("prefix_route", "exit", 1) Syslog output: Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:122]: db_cmd(): DB driver cluster does not implement mandatory db_cmd function Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:168]: db_cmd(): db_cmd: Cannot create db_cmd structure Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: prefix_route [prefix_route.c:126]: pr_db_load(): prefix_route: db_load(): db_cmd() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: : prefix_route [prefix_route.c:214]: mod_init(): prefix_route: db_load() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [sr_module.c:974]: init_mod(): Error while initializing module prefix_route (usr/local/lib64/kamailio/modules/prefix_route.so) Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Ubuntu 14.04LTS version: kamailio 4.3.6 (x86_64/linux) 105a17 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 105a17 compiled on 10:06:12 Oct 4 2016 with gcc 4.8.4 Thank you.

2 3

Enabling accounting module
by Diogenes Marcano 12 Oct '16

12 Oct '16

Can I enable accounting modules acc and acc_cdrs with out diameter? Best regard

2 1

Permissions module question
by Ivan Dudko 11 Oct '16

11 Oct '16

Hello! I am new to kamailio and trying to use it vanilla config. Now main question is how to use ip based auth. I found recent post of Daniel-Constantin Mierla: http://lists.sip-router.org/pipermail/sr-users/2011-December/071147.html Here he recommends to use 'address' table from permissions module, I try yo use advice and add this lines at config begin: #!define WITH_MYSQL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_AUTH #!define WITH_IPAUTH #!define WITH_USRLOCDB Database is created and kamailio can access it. I am add user 1000 to kamailio via kamctl and successful register it with soft-phone. I create trunk without registration to kamailio on asterisk server. And trying to call from asterisk to user 1000. Call is successful. I try to create file /etc/kamailio/permissions.deny with content 'ALL : ALL'. And retry previous call. It still sucessful. I try to add record with asterisk address to 'address' table with group 1. And retry previous call. It still sucessful. I am confused. I do not now how to disable any address for ip_auth except if it in the 'address' table. And allow any address with if it request kamailio with registration. -- Best Regards, Ivan Dudko

3 4

Kamailio - DMQ dns srv Question
by José Seabra 11 Oct '16

11 Oct '16

Hello, I have a doubt related with DMQ dns behavior, I noticed that when kamailio starts, it tries to resolve DMQ name configured on parameter notification_address as the following sequence: 1. SRV 2. A 3. AAAA Isn't supposed kamailio try first resolve the NAPTR DMQ name, and then SRV? I'm asking this because kamailio is trying resolve the SRV record without any transport protocol specified on query, as my dns server only accepts queries on the format "_Service._Proto.Name" the SRV will never be resolved. Thank you. BR -- José Seabra

4 13

transaction is not suspended ERROR
by david 11 Oct '16

11 Oct '16

Hello all i'm having some errors sometimes and i'm not able to know why they happen i made a lua configuration script where i create some ESL sockets with several freeswitches (10) (around 500-600 sockets in total, depending on the number os children (8) and listen ips we have (5)), and i use the lua script to send some API to the FSW by a function. No matter what the FSW replies. i dont know why, but i think that having that conn:bgapi($command) in the lua script, makes the issue to happen. theorically, the lua script waits to receive the response from the fsw through the tcp socket, and after that the script continues (with an async_sleep command), but after the API sent, i see (only sometimes, few times) an async_sleep seem to fail when doing the t_continue with the following error WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [20748:225229907] could it be posible that having manye ESL sockets opened in the kamailio server makes this to happen? how many async workers are needed? i have 8 right now is there a way where i can know or correlate the log to a $ci or something known? best regards david

2 3

Kamailio on Redhat 7.2
by kamailio noob 11 Oct '16

11 Oct '16

Hi All, Can anyone please provide me the details that how to install kamailio on Redhat 7.2. I am following the below url guide for installation of Kamailio. https://www.kamailio.org/wiki/install/4.4.x/git when i compile kamailio code I am not able to find certain scripts in /usr/local/sbin Can anyone help regarding this ? Regards, Atif Ramzan.

2 1

Busy return code when a user has an ongoing transaction
by Gholamreza Sabery 10 Oct '16

10 Oct '16

I want to be able to return a Busy code foe a user that is talking to someone else and receives an INVITE for a new call. How can I implement this in Kamailio? Regards

2 2

Order of record-route
by Dirk Teurlings - Signet B.V. 10 Oct '16

10 Oct '16

Hi, Looking at an issue with SIP-Cisco and a Barracuda firewall, I've come across something I need to know more about. The setup is as follows: CLIENT -> KAMAILIO -> ASTERISK CLIENT <- KAMAILIO <- ASTERISK Client sends INVITE, Kamailio forwards to Asterisk, SIP 100 and SIP 183 are being sent and finally a 200 OK. We now expect the Client to send back an ACK packet, but that never reaches us. In this case the SIP 183 is the first packet that contains the full record route of the path, the 200 OK contain the same record-routes being: Record-Route: <sip:192.168.0.200;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:1.2.3.4;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:4.3.2.1;r2=on;lr;did=4c8.3f471377> Record-Route: <sip:10.0.0.101;r2=on;lr;did=4c8.3f471377> 192.168.0.200 is the private address of Kamailio 1.2.3.4 is the public address of Kamailio 4.3.2.1 is the public address of the Client 10.0.0.101 is some private addres of the Client What we're seeing is that the Client is sending his ACK message to our 192.169.200 private address, which he will never reach. So me question in this case is; is the order of the Record-route headers relevant and do I have any say in which order Kamailio outputs these headers? To be clear, the Kamailio config only contains one reference to record_route which is caught for INVITE messages, which only fires when all things like NAT and transaction checks haven't already picked it up. Kind regards, Dirk

3 5

kamctl - Error opening FIFO
by Hermann Norpois 10 Oct '16

10 Oct '16

Hello, I have problems with the behaviour of kamctl ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl ul show ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. #my FIFO file ubuntu@ip-172-31-32-10:/etc/kamailio$ ls -l /var/run/kamailio/kamailio_fifo prw-rw---- 1 kamailio kamailio 0 Oct 7 09:01 /var/run/kamailio/kamailio_fifo #FIFO in config ubuntu@ip-172-31-32-10:/etc/kamailio$ grep fifo kamailio.cfg loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") #With sudo it works ubuntu@ip-172-31-32-10:/etc/kamailio$ sudo kamctl ul show Domain:: location table=1024 records=0 max_slot=0 Further ... ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' #But kamailio is running ubuntu@ip-172-31-32-10:/etc/kamailio$ ps axw|egrep kamailio 4122 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4123 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4124 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4125 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4126 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4127 ? S 0:46 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4128 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4129 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4130 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4131 ? S 0:03 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4132 ? S 909:27 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4133 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4134 ? S 5:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4135 ? S 25:51 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4136 ? S 26:02 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4137 ? S 26:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4138 ? S 26:18 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4139 ? S 10:17 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 27344 pts/1 S+ 0:00 egrep --color=auto kamailio Whats wrong with my usage of kamctl? Thanks Hermann

3 2

Q: (markup) language for defining structures and function prototypes
by Daniel-Constantin Mierla 10 Oct '16

10 Oct '16

Hello, writing to see if anyone is aware of any (well known) markup language for defining structures and function prototypes. The goal is to use it for specifying the functions exported by KEMI (https://www.kamailio.org/wiki/devel/config-engines) so we can generate C code for the exported structure, but more important, to get at least the index with the functions available over the KEMI. I feel it's going to be a big burden to write C code and docs in two places, even worse, to keep both of them sync'ed over the time. At this moment, there are many functions exported via KEMI, but not documented anywhere (apart of the C code). It should allow defining the name of the function, the parameters with their associated types, return code and eventually the description for each of them. At the end we could invent one based on xml format (or something similar), but I would better use something that already exists there. I searched quickly, but couldn't find anything straightaway. A hint could be a format that is used to document the RPC commands. Once the format is decided, we may have to write a tool to generate C code to be compiled with the module and the xml docbook file to be included in the documentation. Any idea to solve this welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

1 0

Kamailio stop processing SIP
by Igor Potjevlesch 07 Oct '16

07 Oct '16

Hello, Someone has an idea? Again, Kamailio stop processing SIP during more than 30s. Nothing print in the logs. The processing back on when a script detect no answer and restart the service. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : vendredi 23 septembre 2016 11:05 À : sr-users(a)lists.sip-router.org Objet : Dimension number of child Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

3 6

Migrating from OpenSER config file question.
by Brian McCrary 07 Oct '16

07 Oct '16

Hello group, I know this may be hard to believe but I'm in the process of upgrading an old, but stable, OpenSER 1.0 group of servers to Kamilio. I'm basically going to sort of start from scratch with the database and export my old MySQL database and write some scripts to reimport the data using kamctl scripts since the database structure has changed substantially since then it looks like. I've mostly migrated the config file over fairly easily, except for AVPops. I've got a section of config file that writes into the usr_preferences database for call forwarding, but I seem to get the same error anywhere I try to write to the database using AVPops. Here's the output I get when checking the config: 0(1378) DEBUG: <core> [pvapi.c:268]: pv_cache_add(): PV cache not initialized, doing it now 0(1378) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error searching pvar "avp" 0(1378) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:callfwd)] at [5 (5)] 0(1378) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 145, column 4-18: Can't get from cache: $avp(s:callfwd) ERROR: bad config file (1 errors) Here's what I think are the revelant portions of the config file: modparam("avpops", "db_url", "mysql://user:pass@localhost/kamailio") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") <snip> if (avp_db_load("$from/username", "$avp(s:callfwd)")) { if(!avp_db_delete("$from/username", "$avp(s:callfwd)/usr_preferences")) { sl_send_reply("500", "Server DB error"); } } # avp_write("$ruri","$avp(s:callfwd)"); # Old OpenSER 1.0 syntax... $avp(s:callfwd) = $ruri; # New Kamailio syntax ?? avp_db_store("$from/username","$avp(s:callfwd)/usr_preferences"); t_relay(); return; Basically I just want to write the ruri into the usr_preferences database with an attribute of callfwd but not sure what I'm missing. I see the error "Can't get from cache" but not sure how that pertains to what I'm trying to do. Anyone have any ideas? Thanks, Brian

2 5

Size of passwords fields in db
by Daniel-Constantin Mierla 06 Oct '16

06 Oct '16

Hello, writing here to decide on a topic opened by pull request 779: - https://github.com/kamailio/kamailio/pull/779 what would be a fair size for db column storing a password that one would like to have for proper security? I would like to make it consistent over all tables that have a password column by defining a xml entity for the size of these columns. The pull request suggests 64 chars, has anyone other opinions on making it larger or smaller? If they are defined varchar, then should not be a problem of allocated size, so we can go with 128 if someone feels it worth doing larger now so we don't have to change it again in the near future. This change is about db schema, the modules I expect to work with allocated strings (or have length checks) in this case and should not be affected. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

2 4

Fwd: TOPOH Module in Kamialio
by Ginti Saurabh 06 Oct '16

06 Oct '16

Hi, Can you let me know what is TOPOH module do in kamailio? what does it do? and if commented out this line then will it effect the system?

2 6

Load balancing with Kamailio
by Nitesh Bansal 06 Oct '16

06 Oct '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

2 1

Serial Forking and t_suspend
by Kelvin Chua 06 Oct '16

06 Oct '16

we have serial forking already setup properly but need to t_suspend(). found out that it completely breaks the destination set. other than manually rebuilding the destination set after t_continue(), are there any other solution out there? Kelvin Chua

2 1

Re: [SR-Users] Identical Kamailio setup on 2 hosts
by Daniel Tryba 05 Oct '16

05 Oct '16

On Fri, Sep 30, 2016 at 10:29:16PM +0530, Infinicalls Infinicalls wrote: > Hi Daniel, > This seems to be ideal setup for my requirement. Would you mind > sharing the kamailio.cfg file with me (with all the important data > masked) ? Not much important stuff to censor. LBIPADDR is the ipadress of the machine itself. REGISTRARNET/30 is the net of the backend machines. The dispatcher.list contains the backend registrars The backend is essentially the default config (with an always on rtpengine) with the following additions: You need to enable path.so and set modparam("path", "use_received", 1) Tell the registrar module to use path (path_mode 1 fixes you 420 problem): modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) If you want to send keepalives from the registrar to the enduser: modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "udpping_from_path", 1) the loadbalancer config has a little blurb about relaying the sipping from the backend manually using the path header. But I'd advice to ping from the an external process to have better control (eg sipsak)

2 3

Addressing a distinct PSI result is: 555 AS Contacting Failed - iFC terminated dialog
by Valentin Christoph 05 Oct '16

05 Oct '16

Hi all, I'm trying to send a call to an AS via a kamailio S-CSCF (release 4.4.3). I'm addressing the AS by a distinct PSI: sip:04325432101@domain.net;user=gsmr. At the first try the S-CSCF sends a SAR to HSS and receives a successful SAA (tshark traces attached), but the S-CSCF rejects the call with 555 AS Contacting Failed - iFC terminated dialog. At the second try (a few seconds later), the S-CSCF doesn't exchange SAR/SAA with the HSS, and now the call is successfully forwarded to the AS. This problem is permanent, it is not the result of a temporary loss of connectivity between S-CSCF and AS. I checked the iFCs that I had configured at the HSS and I think, they are OK. I attached - tshark traces from the node, where the S-CSCF resides: 2016_09_19_error_555_17_scscf_filtered.pcap - WITH_DEBUG logs from kamailio: 2016_09_19_error_555_17_kamailio_1st_try.log - A tarball of /usr/local/etc/kamailio: 2016_09_19_error_555_17_kamailio.tar.gz Can anybody help? Is it a bug in kamailio or is the problem on my side? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 4

484 Address Incomplete error
by Infinicalls Infinicalls 05 Oct '16

05 Oct '16

Hi all, I am trying a simple proxy (with dispatcher) and a registrar. However, I have this issue U 2016/10/05 08:22:44.057195 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK8653.a9c65320000000000000000000000000.0;received=52.233.25.164. To: <sip:51.140.187.167:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b79c. From: <sip:calls@infinicalls.com>;tag=533cb9e91f4b999cf76861cbb9ed54ed-d3d8. CSeq: 10 OPTIONS. Call-ID: 28fcdbee608363cf-5442(a)127.0.0.1. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. I have set modparam("dispatcher", "flags", 2) Any idea? Thanks. regards Ganesh Kumar

2 1

NatHelper - Received adding on Contact header
by Yasin CANER 04 Oct '16

04 Oct '16

3 4

Sip Proxy server_kamailio Features
by Surender Singh 03 Oct '16

03 Oct '16

Hello, Thanks to all to develop the such type Sip Proxy server. Please suggest whether SIP proxy server is open source ,If yes than it can fulfil the my below requirement . 1) I having some sip clients which are registered on IMS .Can I use the kamailio Sip proxy sever between sip client and IMS. Sip client ------------------Kamailio Sip Proxy ---------------IMS 2) Second things, can I use Kamailio Sip Proxy to interconnect two IMS nodes . Can I create the SIP trunk without registration mode (username and password) between IMS nodes using Kamailio. Because the my IMS node doesn't support sip trunk in registration mode. Its support one normal trunk (SIP-T) My current IMS set up like below. IMS-1 (IBCF)---------------IMS2(IBCF). Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio number of concurrent calls
by Gholamreza Sabery 03 Oct '16

03 Oct '16

I used SIPp to stress test different scenarios in Kamailio, but how can I simulate a real call with media? SIPp has media sending capabilities but in not enough for example to simulate 1000 calls persecond! How can I test this?

5 7

How to check in Kamailio if a current sip message 200 OK is a response to the BYE.
by AFRICALLSHOP 01 Oct '16

01 Oct '16

Hello, I try to check in Kamailio if a current sip message "200 OK" is a response to a "BYE" request. Could you confirm the following code please? IF (IS_METHOD("BYE") && STATUS == 200) { } Regards Abdoul.

2 3

Proxy unable to connect - 420 Bad Extension error
by Infinicalls Infinicalls 01 Oct '16

01 Oct '16

Hi, I have setup a proxy and able to connect to the registrar. However, after a few attempts, it throws the "420:Bad Extension" error, saying "Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated." Any idea, how to overcome this ? 10.1.0.4 is not my IP anywhere. I don;t know where it takes that value from. Thanks. regards Ganesh Kumar

2 2

Call forking priority
by Igor Potjevlesch 30 Sep '16

30 Sep '16

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so I relay first to the fixed proxy, which will bounce back to mobile proxy the branch for mobile device. But now if I want to be able to choose which phone (mobile or fixe) rings before I have tried by doing the "async_route("RELAY", "7");" in the fixe proxy like in the mobile proxy but it doesn't work. My code is: Fix proxy: if($rU==123456789){ add_diversion("forking"); $var(priority)=1; append_branch("sip:987654321@proxy_mobile"); } route[INVITE]{ if($var(priority) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } Mobile proxy: if($tU==987654321){ $rU=123456789; } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } Thanks.

2 2

ims_charging state machine issue
by 4fba81+8i7felu07050＠guerrillamail.com 30 Sep '16

30 Sep '16

Wanted to ask about changing rates while a call is not yet connected. I did this, $var(cc_ret1) = Ro_CCR_Stop("term", "487", "Request Terminated"); $var(cc_ret2) = Ro_CCR("RO_REPLY", "term", 30, "1", "1"); however, upon sending the new CCR, i get this ERROR: cdp [acctstatemachine.c:185]: cc_acc_client_stateful_sm_process(): Received unknown event [1] in state [1] and CRITICAL: ims_charging [ro_timer.c:122]: insert_ro_timer(): Trying to insert a bogus ro tl=0x7f484ab48728 tl->next=0x7f484ab42858 tl->prev=0x7f484aaa1090 looks like there is some statemachine issue. question is, how do we change rate? ---- Sent using GuerrillaMail.com Block or report abuse: https://www.guerrillamail.com/abuse/?a=UlM5EB4FUac1hAen5G1RYR3XANifx8NV1epN…

1 0

Identical Kamailio setup on 2 hosts
by Infinicalls Infinicalls 30 Sep '16

30 Sep '16

Hi, I have two identical hosts running Kamailio with the same set of users. They are located in different locations. Both are running behind NAT and I've enabled rtpproxy for that and also advertised the public IP. Both the MySQL DBs are similar and have the same data. Both the hosts have different sub-domains in alias and I have set auto_alias=no Now, the problem is my users are not able to communicate from one server to another. user1(a)mydomain.com from host1 is not able to communicate to user1(a)mydomain.com at host2, though inter-server communication is happening nicely. Any idea how to solve this? Thanks. regards Ganesh Kumar -- --- http://www.infinicalls.com

3 8

Load balancing traffic with Kamailio
by Nitesh Bansal 29 Sep '16

29 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the performance of the Asterisk ,it would be great if we could take a look at SIP response time from Asterisk and use that as metric for load balancing the traffic) Is there any Kamailio module which could do that? Do I need to integrate some other tool with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

3 3

Re: [SR-Users] Distributed Message Queue Module
by Charles Chance 29 Sep '16

29 Sep '16

Hello, That depends entirely on what you need to replicate between servers. DMQ is just a communication channel. How you use it is up to you. For example, if you enable dmq_usrloc, there's no need to share a location database. Htable has an option for replicating over dmq too. The number of direct module integrations like dmq_usrloc/htable is fairly limited, but you can also share arbitrary data between nodes yourself from within your config. I think you'll need to provide a little more information about what you're trying to achieve, however, in order for us to offer any further advice. Cheers, Charles On 29 Sep 2016 08:08, "Moacir Ferreira" <moacirferreira(a)hotmail.com> wrote: Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

1 0

Distributed Message Queue Module
by Moacir Ferreira 28 Sep '16

28 Sep '16

Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir

1 0

memory problem
by beer Ll 28 Sep '16

28 Sep '16

hi all I have kamailio 4.2.0 on Debian Jessie the parameters on startup of kamailio are : SHM_MEMORY=512 PKG_MEMORY=32 the server is dedicated to the kamailio SIP service with authentication and accounting on a separated db server the server process about 30/50 call setups per second everythings works good but after 5/6 days the processes lock and I received a lot of error log like these: <core> [rvalue.c:1710]: rval_str_add2(): rv eval out of memory <core> [msg_translator.c:362]: received_builder(): out of memory <core> [msg_translator.c:2717]: via_builder(): out of memory ... <core> [msg_translator.c:2423]: build_res_buf_from_sip_req(): out of memory; needs 514 ... <core> [data_lump.c:172]: insert_new_lump_before(): out of memory acc [acc_extra.c:267]: extra2strar(): extra2strar: out of memory. after a kamailio reboot evrethings works again for 5/6 days. what can I do to resolve?

3 3

RPC tcp limit buffer
by pars3c 28 Sep '16

28 Sep '16

Hello, i'm trying to list the dialog (dlg_profile list) in the proxy using an RPC call to jsonrpc-s (dlg_profile list). But when there are many calls , kamailio send an error because the tcp buffer is not big enough. The only solution is to increase the parameter tcp_conn_wq_max every time i've an error ? Thanks

2 2

Accessing cached database table
by Daniel W. Graham 28 Sep '16

28 Sep '16

Is there a way to access a cached database table from the config file, without using a function from a module? Example - Lookup ip_addr from 'address' table or src_ip from 'trusted' table based on the 'tag' column. No function exists in permissions module to do this, but running 'kamctl address show ' or 'kamctl trusted show ', show that the tables along with tag column are cached. Thanks! -Dan

2 2

invalid sendtoparameters
by Linux Vince 27 Sep '16

27 Sep '16

I am getting this error on every outgoing request. Below are log entries i found Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7fb76f5da5e0,1568,0, 10.3.4.2:5060,16): Invalid argument(22) Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [forward.h:201]: msg_send_buffer(): udp_send failed SNDTO values are SNDTO ,10.3.4.2 PORT 5060 I have confirmed the PID and it is bound to my public IP 22417 main process - attendant 22418 udp receiver child=0 sock=10.3.4.1:5060 22419 udp receiver child=1 sock=10.3.4.1:5060 22420 udp receiver child=2 sock=10.3.4.1:5060 22421 udp receiver child=3 sock=10.3.4.1:5060 22422 udp receiver child=4 sock=10.3.4.1:5060 22423 udp receiver child=5 sock=10.3.4.1:5060 22424 udp receiver child=6 sock=10.3.4.1:5060 22425 udp receiver child=7 sock=10.3.4.1:5060 22426 udp receiver child=0 sock=127.0.0.1:5060 22427 udp receiver child=1 sock=127.0.0.1:5060 22428 udp receiver child=2 sock=127.0.0.1:5060 22429 udp receiver child=3 sock=127.0.0.1:5060 22430 udp receiver child=4 sock=127.0.0.1:5060 22431 udp receiver child=5 sock=127.0.0.1:5060 22432 udp receiver child=6 sock=127.0.0.1:5060 22433 udp receiver child=7 sock=127.0.0.1:5060 22434 slow timer 22435 timer 22436 secondary timer 22437 ctl handler 22438 MI FIFO 22439 WEBSOCKET KEEPALIVE 22440 tcp receiver (generic) child=0 22441 tcp receiver (generic) child=1 22442 tcp receiver (generic) child=2 22443 tcp receiver (generic) child=3 22444 tcp receiver (generic) child=4 22445 tcp receiver (generic) child=5 22446 tcp receiver (generic) child=6 22447 tcp receiver (generic) child=7 22448 tcp main process How do i identify the issue ?

1 1

SIREMIS MANUAL
by Carlyle Edwards 27 Sep '16

27 Sep '16

Hi, is there a manual for SIREMIS?

2 1

Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF
by Owais 26 Sep '16

26 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-IMS-NAT-issue-PRACK-ACK-RU… Sent from the Users mailing list archive at Nabble.com.

2 2

REGISTER between Kamailio 4.4 and Asterisk 13
by Grant Bagdasarian 26 Sep '16

26 Sep '16

Hello, We're trying to get our Asterisk test server to REGISTER with Kamailio. The first attempts are ok, but once the nonce timer is exceeded Kamailio starts rejecting the REGISTER requests from Asterisk, due to the nonce being expired. The auth_check function is the one returning error -4. Is this an issue in Asterisk or Kamailio? Which component is responsible for generating a new nonce value? I'm a bit confused regarding the flow of SIP REGISTER. I hope someone could help me out on this. Thanks! Regards, Grant Bagdasarian

2 2

Config changes needed from 4.4.2 to 4.4.3 (and a bug)
by Sebastian Damm 26 Sep '16

26 Sep '16

Hi, Kamailio release 4.4.3 was tagged as a "bug fix release", no config changes should be needed. However, we had to change two things in our config file to get 4.4.3 running. I thought, I might share those on the list. 1) Defines must not have '-' in the names. In 4.4.2 we had a Define like this: #!define NO-FRAUDCHECK This threw an error when trying to restart with 4.4.3. Replacing it with #!define NO_FRAUDCHECK fixed the problem. 2) Module Order problems. We introduced the usage of http_async_client module a few weeks ago, and this worked perfectly. However, 4.4.3 complained that the pv module was not available before loading http_async_client. After moving up the pv module above http_async_client, everything worked again. 3) There's a bug in uac_auth function. I've opened an issue for that: https://github.com/kamailio/kamailio/issues/797 Best Regards, Sebastian

2 1

Textops replace question
by Igor Olhovskiy 26 Sep '16

26 Sep '16

Hi! I’m trying to use replace function, but got strange results. Idea - modify Contact header What I got as start packet. (202 Accepted on onreply_route) … Contact: <sip:*97@10.0.20.81:5060> … I give a command replace("Contact: <.*>», "Contact: <" + $fu + ">"); $fu = sip:11@master.rufan.at And, as a result I got Contact: <sip:11@master.rufan.at>sip:*97@10.0.20.81:5060 Why it’s adding old Contact part at the end? What can cause this? On onreply_route I have only this command. Thanks! -- Best regards, Igor

3 4

Dialog sharing between Kamailio servers
by Michael Jepson 26 Sep '16

26 Sep '16

Hi all, I am trying to limit the concurrent calls of a customer, but we have multiple Kamailio servers in our platform. I have the dialog module up and running and it is storing its data in a database, but when I use get_profile_size, it will only return the number of dialogs in that profile for the current machine, it does not add the others. How can I get Kamailio to add the others, i.e. how can I get Kamailio to consult the database to get the correct size. Now, Kamailio seems to only write to the database, but not read from it. Best regards, MICHAEL JEPSON

2 1

Load balancing with Kamailio
by NITESH BANSAL 26 Sep '16

26 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome.

3 3

LCR routing, and finaly route to media gateway if not found
by Loic Chabert 26 Sep '16

26 Sep '16

Hello List, I'm finishing my implementation of LCR with kamailio for outgoing calls but one use case does not work. Let me explain: - I have some sbc for one destination. If one destination fail, kamailio try to second one etc.. - If no SBC is available for this destination (for exemple, all are down), i want to send my call to an error server (asterisk), play an error sound. To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", and launch RELAY route. But kamailio dont make a DNS query to find IP of "errorserver.local", only change RURI and try to send to the last LCR gateway. I have also try to add a new branch, but same result... Why kamailio do not make a new DNS query ? *route[RELAY] { #if (is_method("INVITE")) { # if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); #} if (!t_relay()) { sl_reply_error(); } exit;}route[NOTFOUND]{ $ru = "sip:error1@errorserver.local"; xlog("L_WARN", "[$ci]|$si|Routing this call to $ru (error server)\n"); append_branch($ru); route(RELAY);}route[LCR] { xlog("L_INFO", "[$ci]|$si|Loading gateways...\n"); if (!load_gws(1)) { xlog("L_ERR", "[$ci]|$si|Can't load LCR gateways\n"); sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO", "[$ci]|$si|This gateway was found: gw_uri_avp[$var(i)]=$(avp(i:709)[$var(i)]) \n"); $var(i) = $var(i) + 1; }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_INFO", "[$ci]|$si|No more gateways. Goto route NOTFOUND...\n"); route(NOTFOUND); }; # try the first matched gateway if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; }; exit;}failure_route[lcrfailure] { # the previous gateway is no good if (t_check_status("408|50[34]")) { xlog("L_WARN", "[$ci]|$si|Entering in entering failure_route[lcrfailure]. Error code: $T_reply_code \n"); if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_ERR", "[$ci]|$si|AVP is not set\n"); }; if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; exit; };}* Many thanks, Regards.

3 2

how to config gateway in kamailio
by Gaurav Bmotra 26 Sep '16

26 Sep '16

Hi i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway -- *Regards:* Gaurav Bmotra

2 1

Scripting Issue
by aman jeff 26 Sep '16

26 Sep '16

Hai Team, I can't able to set maxptime in SDP Please help me

1 1

Location Table In db_mode 3
by Gholamreza Sabery 25 Sep '16

25 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

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