- sr-users - kamailio.org

I-CSCF Prefixes IMPU with + Symbol
by Ron McLeod 25 Oct '17

25 Oct '17

I am working with v5.0.3 and the IMS extensions. When I attempt to send a SIP MESSAGE from the application server towards the UE, the I-CSCF used to determine the S-CSCF for the subscriber by sending a Location-Info request to the HSS. I found that this request was failing because the IMPU (IMS Public User Identity) was being mangled by the I-CSCF, prefixing the user part with a + symbol. The example configuration file for the I-CSCF does in-fact does cause the r-uri to be prefixed with a + symbol if one is not already present. I'm wondering why since it breaks the LIR/LIA lookup. AS I-CSCF HSS | MESSAGE | | | ---------------------> | | | sip:ron@provider1.test | | | | | | | | | | LIR | | | ----------------------> | | | sip:+ron@provider1.test | | | | | | | | | DIAMETER_UNKNOWN_USER | | | <---------------------- | | | | | | | | 403 | | | <--------------------- | | | FORBIDDEN | | route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } ... Thanks, Ron

1 0

Testing for user=phone with siputils uri_param() doesn't work
by George Diamantopoulos 25 Oct '17

25 Oct '17

Hello, In my script I've had to test for the user=phone parameter in URIs. I thought uri_param() from siputils would be handy for this operation on the RURI, so I did: if ( !uri_param("user","phone") ) { add_uri_param("user=phone"); } Unfortunately this wouldn't work (uri_param returns -1), and I end up with double user=phone params in the RURI. I was able to work around this by changing the test to: if ( !($(ru{uri.uparam}) == "phone") ) { add_uri_param("user=phone"); } But I wanted to ask if this is intentional or if I should file a bug or something. Thanks. George

3 3

Planning Kamailio v5.0.4
by Daniel-Constantin Mierla 25 Oct '17

25 Oct '17

Hello, I am considering to release Kamailio v5.0.4 later this week, likely on Wednesday or Tuesday. If you are aware of any issue in branch 5.0 which is not yet reported on github.com/kamailio/kamailio issues tracker, then do it as soon as possible to have a chance to look at it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 5

usrloc module does not work with db_cluster
by Uri Shacked 25 Oct '17

25 Oct '17

Hi, I started using db_cluster module. DB are master-slave replication both ways. #!define DBURL "cluster://kamailio" when i stop the primary mysql server, all subscribers fails to register again. seems like the usrloc module does not work with the cluster. modparam("usrloc", "db_mode", 1) modparam("db_cluster", "cluster", "kamailio=>primary=9s9s;secondary=8s8s") it works OK for modules like dispatcher and htable to reload from DB. it shows the sql driver errors, but reloads from the secondary server. I need auto expire, so, p_usrloc is not a good option. working with version 4.2 any ideas ? Uri

2 1

FW: Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 25 Oct '17

25 Oct '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

1 0

Advice on taking the best path to build a Presence & IM server
by José Seabra 24 Oct '17

24 Oct '17

Hello there, I'm writing here in order to get your experiences about a Presence & IM server. I would like to know what is the best solution to follow: - SIP Simple or XMPP? I've more knowledge about SIP Simple with kamailio but i did some researches and i found several open sources unified communications software that's implement Presence and IM using an XMPP server. So what is your opinion about which is the best solution to choose, taking in considerations the following aspects: 1. scalability 2. interopeability 3. permornace 4. features(like: Personal assistance, voicemail integration, chat group, integration with social media) Should i choose Kamailio SIP Simple with xcap or an XMPP server(Prosody or ejabberd, etc) I will appreciate a lot your opinion. Thank you Regards José -- Cumprimentos José Seabra

3 4

lcr
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

(no subject)
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

How to check 4XX/5XX response in general.
by vivek＠advaitamtech.com 24 Oct '17

24 Oct '17

Hi, Is there any way to catch all 4XX and 5XX SIP error codes rather then specifying individually ie. 404, 500, 400 etc in kamailio. Thanks, Vivek.

2 2

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 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, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

No music on hold with mohqueue
by Fedor 24 Oct '17

24 Oct '17

I got in my kamailio.cfg: loadmodule "mohqueue.so" # --- music on hold moh ---- modparam ("mohqueue", "db_url", "mysql://kamailio:PASSWORDHERE@localhost/ kamailio") modparam ("mohqueue", "mohdir", "/etc/kamailio/moh") modparam ("mohqueue", "moh_maxcalls", 3000) Databases for mohqueue exist rttproxy configured fed@vps1:/etc/kamailio# ll /etc/kamailio/moh/ total 185424 drwxr-xr-x 2 kamailio kamailio 4096 Oct 8 09:32 ./ drwxr-xr-x 3 root root 4096 Oct 8 10:04 ../ -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.0 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.1 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.10 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.11 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.12 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.13 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.14 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.15 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.16 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.17 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.18 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.19 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.2 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.20 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.3 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.4 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.5 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.6 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.7 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.8 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.9 -rw-r--r-- 1 kamailio kamailio 8627660 Jan 2 2017 hold.wav rtpproxy configured, but no music on hold(srtp works fine) Can anybody help me, please?

2 1

Multi crashes in the same day
by igor.potjevlesch＠gmail.com 23 Oct '17

23 Oct '17

Hello ! I have multi crashes on a Kamailio 4.2.8 instance. Already 6 coredump today, same time with the same scenario. Below the output of the last coredump: (gdb) where #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 #2 0x000000000061aad2 in qm_debug_frag (qm=0x7f9a8a7aa010, f=0x7f9a8aca5968) at mem/q_malloc.c:161 #3 0x000000000061bf1c in qm_malloc (qm=0x7f9a8a7aa010, size=24, file=0x7131d0 "<core>: re.c", func=0x71483e "subst_run", line=440) at mem/q_malloc.c:388 #4 0x0000000000505e2c in subst_run (se=0x7f9a8ac08b78, input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, count=0x0) at re.c:440 #5 0x0000000000506fff in subst_str (input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, se=0x7f9a8ac08b78, count=0x0) at re.c:515 #6 0x00007f9a87ff629d in subst_uri_f (msg=0x7f9a8abfef38, subst=0x7f9a8ac08b78 "X\215\300\212\232\177", ignored=0x0) at textops.c:744 #7 0x000000000041d51b in do_action (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1094 #8 0x0000000000429ba3 in run_actions (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1583 #9 0x000000000042a208 in run_actions_safe (h=0x7fffed081380, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1648 #10 0x0000000000542994 in rval_get_int (h=0x7fffed081380, msg=0x7f9a8abfef38, i=0x7fffed080890, rv=0x7f9a8ab64ef8, cache=0x0) at rvalue.c:924 #11 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed081380, msg=0x7f9a8abfef38, res=0x7fffed080890, rve=0x7f9a8ab64ef0) at rvalue.c:1918 #12 0x000000000041cf77 in do_action (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1064 #13 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1583 #14 0x000000000041d3f6 in do_action (h=0x7fffed081380, a=0x7f9a8ab6b070, msg=0x7f9a8abfef38) at action.c:1079 #15 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab607b8, msg=0x7f9a8abfef38) at action.c:1583 #16 0x000000000042a2d0 in run_top_route (a=0x7f9a8ab607b8, msg=0x7f9a8abfef38, c=0x7fffed081380) at action.c:1669 #17 0x00007f9a891f6d0f in prepare_new_uac (t=0x7f9a6de86518, i_req=0x7f9a8abfef38, branch=0, uri=0x7fffed0814d0, path=0x7fffed0814b0, next_hop=0x7f9a8abff1a8, fsocket=0x0, snd_flags=..., fproto=0, flags=2, instance=0x7fffed0814a0, ruid=0x7fffed081490, location_ua=0x7fffed081480) at t_fwd.c:410 #18 0x00007f9a891faded in add_uac (t=0x7f9a6de86518, request=0x7f9a8abfef38, uri=0x7f9a8abff1a8, next_hop=0x7f9a8abff1a8, path=0x7f9a8abff568, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0x7f9a8abff578, ruid=0x7f9a8abff590, location_ua=0x7f9a8abff5a0) at t_fwd.c:855 #19 0x00007f9a89201ddd in t_forward_nonack (t=0x7f9a6de86518, p_msg=0x7f9a8abfef38, proxy=0x0, proto=0) at t_fwd.c:1723 #20 0x00007f9a891ee374 in t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, proto=0, replicate=0) at t_funcs.c:355 #21 0x00007f9a8922bd89 in _w_t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, force_proto=0) at tm.c:1517 #22 0x00007f9a8922ceee in w_t_relay (p_msg=0x7f9a8abfef38, _foo=0x0, _bar=0x0) at tm.c:1718 #23 0x000000000041d48d in do_action (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1088 #24 0x0000000000429ba3 in run_actions (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1583 #25 0x000000000042a208 in run_actions_safe (h=0x7fffed0848f0, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1648 #26 0x0000000000542994 in rval_get_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, i=0x7fffed082668, rv=0x7f9a8a83eb60, cache=0x0) at rvalue.c:924 #27 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082668, rve=0x7f9a8a83eb58) at rvalue.c:1918 #28 0x0000000000546fc2 in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082af0, rve=0x7f9a8a83f280) at rvalue.c:1926 #29 0x000000000041cf77 in do_action (h=0x7fffed0848f0, a=0x7f9a8a83fbd0, msg=0x7f9a8abfef38) at action.c:1064 #30 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a838c88, msg=0x7f9a8abfef38) at action.c:1583 #31 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996598, msg=0x7f9a8abfef38) at action.c:712 #32 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a9955b8, msg=0x7f9a8abfef38) at action.c:1583 #33 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996a50, msg=0x7f9a8abfef38) at action.c:1079 #34 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a8fa068, msg=0x7f9a8abfef38) at action.c:1583 #35 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:712 #36 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:1583 #37 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830db8, msg=0x7f9a8abfef38) at action.c:1079 #38 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38) at action.c:1583 #39 0x000000000042a2d0 in run_top_route (a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38, c=0x0) at action.c:1669 #40 0x00000000005095b6 in receive_msg ( buf=0xa78b60 "INVITE sip:0123456789@A.B.C.181:5060;transport=UDP;user=phone SIP/2.0\r\nFrom: <sip:0987654321@A.B.C.148:5060>;tag=BN1507883839-1-1507047459-332794206;tran sport=UDP\r\nTo: <sip:0123456789@H.G.F.D"..., len=1119, rcv_info=0x7fffed084be0) at receive.c:216 #41 0x000000000060945a in udp_rcv_loop () at udp_server.c:521 #42 0x00000000004a6e62 in main_loop () at main.c:1629 #43 0x00000000004acfa6 in main (argc=7, argv=0x7fffed085018) at main.c:2581 Regards, Igor.

2 2

Presence WARNING after publish and no NOTIFY
by Yuriy Gorlichenko 23 Oct '17

23 Oct '17

Hi use 5.0.1 with presence for TCP/UDP endpoints presence works PERFECT but for WS socket presence i see next (small cleaned log here with all described bellow https://pastebin.com/8a77ggj0 ): When WS endpoint subscribes - kamailio After kamailio sends 202 to SUBSCRIBE message (see it on client) it sends fast NOTIFY correcly without any trouble, and i see at the kamailio reply from WS endpoint - 200 WS endpoint added to active_watchers table. But when Subscribed endpoint makes for example some call and presence server sends PUBLISH i see: presence [presentity.c:426]: delete_presentity_if_dialog_id_exists(): Presentity already exists - deleting it At the logs and then presence server doesn't send NOTIFY to WS endpoint What is the issue of this? Can not understand why it happens. Can someone to help me understand this? And the dump i see that kamailio send first NOTIFY form UDP port but then translates it to tls for websocket and i see correct NOTIFY at the WS endpoint

2 4

MoS(QoS) measurement in Kamailio based IMS network
by Kranti Kumar 23 Oct '17

23 Oct '17

Hi All, I am using Kamailio IMS VMimage which is provided by Franz. I am successfully established the call between the IMS clients. Here I would like to measure the MoS mean opnion score (QoS) of Calls (Voice quality). Can any one suggest me / guide me how to measure the same in this IMS network. Thank you so much in advance. Regards, -kranti

2 1

how to bridge two landline users
by 赵国杰 23 Oct '17

23 Oct '17

Hello, In a standard sip flow, the call goes like: sip user A --> kamailio --> pstn --> landline user B. However, when user A has a bad internet access, the audio is broken. So what I want is to let sip user A send a invite to kamailio first, then kamailio send invite to user A and B's landline number through pstn, then bridge the two call together. I understand this can be achieved by using FREESWITCH originate and bridge command. I've tried but there's no audio both ways, which really makes me feel stupid of myself. So I'm wondering if this can be done with kamailio? If so, how? Thanks Jesse

2 1

uac_replace_from and header adulterations
by Alex Balashov 23 Oct '17

23 Oct '17

Hello, I've got a bizarre problem caused by bad UA behaviour: UA A ---> Kamailio (P) ---> UA B 1. UA A sends initial INVITE through P to B; 2. Kamailio (P) makes some modifications to the From header using uac_replace_from() and passes along to B. 3. B sends an in-dialog request (e.g. BYE or reinvite) to A through P; in doing so, it modifies the To (formerly From) value slightly, replacing the hostname portion in the To URI with a different value to the one that was received in the From header. 4. Kamailio relays this in-dialog request to A, but with an adulterated/clipped/truncated/grammatically invalid To header now. 5. A responds with 400 Bad Request due to invalid To header. I would not dispute that UA B should not be modifying the remote URI in this manner. But since it does, it gives rise to two questions: 1. Why does Kamailio not simply discard the modified To header and restore the original value stored in the Record-Route rider parameter? Is it because the Record-Route parameter does not contain the original header value, but rather some data complementary to the current header value? Or is it because the UAC code takes a checksum of the original remote URI header value and stores it, and just checks it when restoring on principle? If so, what's the motive for that? 2. Wouldn't it be better behaviour to simply reject a request so malformed, rather than passing it on with a corrupt restored value? If Kamailio can detect that the header has been tampered with, why pass it on? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Help on Video Conference
by Kranti Kumar 23 Oct '17

23 Oct '17

Hello , I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ My configuration details: HSS - 10.0.0.9 (./hss.sh) PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg) ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg) SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg) host machine: 10.0.0.5 I am able to register mobile clients ( IMSdroid) and make voice/video calls between them successfully. Now I would like to make a conference call among 3 IMSdroid users but its failing. Could you please suggest me the way forward here. I am thinking to integrate Application server (AS) to enable Video Conferences. Could you please guide me how to enable AS in this VM image and configuration details. Is there any possibility to do Video conference using this current setup(VM image). Could you please suggest me. Thank you so much in advance. Regards, -kranti

2 1

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

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call_id in dialog module
by ‪mahsa _a_‬ ‪ 20 Oct '17

20 Oct '17

hi all;i want to write "billing" project with kamailio,i using python,postgresql, dialog module and falcon.my problem is in "call_id" section.my code(this section) is in python : """... if (method == 'INVITE'): callid = req.get_param_as_list("call_id") print(req.get_param_as_list("Ts")) print("call id = %s " % callid) elif (method == 'ACK'): time_req = int(req.get_param_as_list("Ts")[0]) callid = req.get_param_as_list("call_id")[0] print("user %s call to %s at time %d % (from_user,to_user,time_req))... """ and in kamailio.cfg: """http_query("http://127.0.0.1:5000/test?method=$rm& call_id=$dlg(callid)&Ts=$Ts&fu=$fu&tu=$tu","$var(result)"); """" "callid" in ACK is ok but "callid' in invite is "null", where is my problem? thanks.

2 1

Retransmission handling
by Volkan Oransoy 19 Oct '17

19 Oct '17

Hi all, I wonder purpose of this block in default script. What I didn’t understand is the purpose of "t_check_trans()”. Accoording to documentation, this method returns a boolean. So why do we use it even we don’t check output? # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } /Volkan

3 3

VoLTE Capable LTE Handsets which support IMS uisng IPv4
by Ron 19 Oct '17

19 Oct '17

I would like to support VoLTE using an IPv4 IMS network rather than IPv6 network. The handsets that I have been trying insist on the IMS network support either IPv6 or IPv4v6 dual stack. After giving them IPv4v6 connectivity, I am finding that even though the handsets asked for both IPv4 and IPv6 P-CSCFs, they only try the IPv6 proxy (if only the IPv4 proxy is provided, they will not try at all). I am wondering of anyone has worked with any LTE handsets which are willing to support IMS using IPv4 only. Thanks, Ron

3 2

Kamailio and FreeSwitch integration
by Timothy oladapo olawuyi 19 Oct '17

19 Oct '17

Dear All, I am a new entrant in Kamailio though little understanding of SIP and FreeSwitch soft switch, however, I have a project to integrate Kamailio and Freeswitch. Here is the scenario of what I want to implement; Kamailio will act as our SIP control while Freeswitch will act media server for incoming calls. Freeswitch will send all outgoing calls to Kamailio for onward transfer to our SIP provider network. No registration, presence, location Accounting, Authentication etc. are required. I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and Kamailio integration sample config available at http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still unable to figure out how to go with the configuration. I will appreciate any one who can provide guideline for achieving the above scenario. Regards Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims | <htttp://www.pethahiah.com/> htttp://www.pethahiah.com

3 4

port forwarding
by Andrzej Kaczmarczyk 19 Oct '17

19 Oct '17

Hello, This is probably trivial, but I'm quite new in kamailio and trying to handle it for my personal purposes (so - no trainings, no support etc :) ) - for few (up to 20) users and rare connections, more to learn something than to have production system. No plans to go outside my own server. I have LAN. One of machines in this LAN is server with kamailio onboard. It works, but only with local hostname (user@hostname). Now I want to go outside (with clients). As temporarty solution I have only dynamic DNS (no-ip.org) and I didn't expect it working. But tried with IP adress (user(a)WAN.IP.ADD.RESS). And... I don't know which ports to redirect on router. Now there are redirected 5060 and 10000-30000 (maybe it is not necessary but first I want it working) - taken from some on-line info. I can: - log in to all my accounts, - depends on client - call to another user and in some cases I can receive the call (so it is probably issue on my clients, because it is possible to make a call at all). But I cannot transmit any data: connections are completely silent in any case and I cannot start video - although cameras are findable. Which ports in fact are necessary? I want audio and video communication, presence info and probably nothing more. Which ports should I redirect? -- Pozdrawiam Andrzej Kaczmarczyk

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 19 Oct '17

19 Oct '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM18/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2018/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 3rd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

1 0

Problems with AAR over Rx Interface
by Ron McLeod 19 Oct '17

19 Oct '17

When a LTE handset client attempts to REGISTER with the IMS, an AAR is sent from the P-CSCF to the PCRF to establish a dedicated bearer for SIP signaling (as well as request notifications for certain access network event). There are a couple of problems with the flow descriptions, and I was hoping that someone could confirm that these are actual issues before I start working on a fix. 1. The "in" and "out" directions seem reversed. "in" corresponds to the uplink (UE to network), and "out" corresponds to the downlink (network to UE). In the AAR sent to the PCRF, the "in" description indicates the traffic "from" the UE, rather than "to" the UE. There is a similar problem with the "out" description. A fix would involve reversing "in" and "out". 2. The IP address and port number for the UE side are correct (in my case the UE has been allocated 172.25.152.195, and the client is using port 37843), but the network side is specified as IP address 127.0.0.1 and port number is the same as the open used by the client (in my case the P-CSCF has an IP address of 172.25.152.20 and a port number of 4060). I see in the source code that the P-CSCF address is hard-coded to "127.0.0.1". A fix would involve either using the actual UE-facing IP address and port number for the P-CSCF or wildcards. Another option would be to not try and setup a dedicated bearer for SIP signaling, and just use the default bearer (there shouldn't be anything else other than signaling-related traffic on the default bearer anyway). The access network HSS would specify that the default bearer was created with the appropriate parameters (QCI=5, AM, etc.). This is the network message flow: 172.25.152.195 -> pcscf.core.ims1.test SIP 999 Request: REGISTER sip:provider1.test (fetch bindings) | pcscf.core.ims1.test -> 172.25.152.195 SIP 359 Status: 100 Trying (0 bindings) | pcscf.core.ims1.test -> pcrf.ims1.test DIAMETER 762 cmd=AARequest(265) flags=RP-- appl=3GPP Rx(16777236) h2h=28fe20b1 e2e=fe338ed9 pcrf.ims1.test -> pcscf.core.ims1.test DIAMETER 198 cmd=AAAnswer(265) flags=-P-- appl=3GPP Rx(16777236) h2h=2 e2e=13 pcscf.core.ims1.test -> 172.25.152.195 SIP 410 Status: 408 Request Timeout (0 bindings) | This is the DIAMETER AAR message: Diameter Protocol Version: 0x01 Length: 696 Flags: 0xc0, Request, Proxyable Command Code: 265 AA ApplicationId: 3GPP Rx (16777236) Hop-by-Hop Identifier: 0x28fe20b1 End-to-End Identifier: 0xfe338ed9 AVP: Session-Id(263) l=40 f=-M- val=pcscf.core.ims1.test;548405219;1 AVP: Origin-Host(264) l=28 f=-M- val=pcscf.core.ims1.test AVP: Origin-Realm(296) l=22 f=-M- val=provider1.test AVP: Auth-Application-Id(258) l=12 f=-M- val=3GPP Rx (16777236) AVP: Vendor-Specific-Application-Id(260) l=32 f=-M- AVP: Destination-Realm(283) l=22 f=-M- val=core.ims1.test AVP: Subscription-Id(443) l=52 f=-M- AVP: Media-Component-Description(517) l=304 f=VM- vnd=TGPP AVP Code: 517 Media-Component-Description AVP Flags: 0xc0 AVP Length: 304 AVP Vendor Id: 3GPP (10415) Media-Component-Description: 00000206c0000010000028af0000000100000207c00000bc... AVP: Media-Component-Number(518) l=16 f=VM- vnd=TGPP val=1 AVP: Media-Sub-Component(519) l=188 f=VM- vnd=TGPP AVP Code: 519 Media-Sub-Component AVP Flags: 0xc0 AVP Length: 188 AVP Vendor Id: 3GPP (10415) Media-Sub-Component: 000001fdc0000010000028af00000001000001fbc0000046... AVP: Flow-Number(509) l=16 f=VM- vnd=TGPP val=1 AVP: Flow-Description(507) l=70 f=VM- vnd=TGPP val=permit out ip from 172.25.152.195 37843 to 127.0.0.1 37843 AVP: Flow-Description(507) l=69 f=VM- vnd=TGPP val=permit in ip from 127.0.0.1 37843 to 172.25.152.195 37843 AVP: Flow-Usage(512) l=16 f=VM- vnd=TGPP val=AF_SIGNALLING (2) AVP: Media-Type(520) l=16 f=VM- vnd=TGPP val=CONTROL (4) AVP: Codec-Data(524) l=28 f=VM- vnd=TGPP val=downlink\noffer\n AVP: Codec-Data(524) l=27 f=VM- vnd=TGPP val=uplink\nanswer\n AVP: Flow-Status(511) l=16 f=VM- vnd=TGPP val=ENABLED (2) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=CHARGING_CORRELATION_EXCHANGE (1) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_LOSS_OF_BEARER (2) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_RECOVERY_OF_BEARER (3) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_RELEASE_OF_BEARER (4) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_ESTABLISHMENT_OF_BEARER (now void) (5) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=IP-CAN_CHANGE (6) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=ACCESS_NETWORK_INFO_REPORT (12) AVP: Framed-IP-Address(8) l=12 f=-M- val=172.25.152.195 AVP: Authorization-Lifetime(291) l=12 f=-M- val=7200 AVP: Auth-Grace-Period(276) l=12 f=-M- val=0 AVP: Session-Timeout(27) l=12 f=-M- val=7200

2 1

Problem removing headers from FAILURE_ROUTE
by Joel Serrano 17 Oct '17

17 Oct '17

Hi all, I have the following scenario: Client -> Kamailio (adds X-CallSource hdr) -> Media Server -> Kamailio (removes all "X-" headers) -> Carrier Carrier doesn't get any X- headers. All works. Now for example, if the media server replies with a non-OK status, like: Client -> Kamailio (adds X-CallSource hdr) <-> Media Server (480 Temporarily Unavailable) Kamailio (in failure_route, removes all "X-" headers) -> External Voicemail Service External Voicemail Service gets the X- headers. Basically, on the first example, it is a regular call, all handled from request route. On the second example, call goes to media server, media server replies with a 480, reaches kamailio's failure route, in there we run the same remove_hf_re. In the captures I can see that the X-CallSource header (added at the very beginning) is still there. I have tried different combinations in the failure route: a) Tried removing exactly the header: remove_hf("X-CallSource") b) Tried removing any X- header: remove_hf_re("X-") Those didn't work, then I tried: c) append_hf("X-CallSource: Test") remove_hf("X-CallSource") The result was now two X-CallSource headers, the add worked, the remove didn't. I thought maybe "msg_apply_changes()" could help, but it isn't valid inside failure_route. So what is the deal here? Am I doing something completely wrong or is it just that you cannot remove headers (but you can add) in failure route? According to the doc: https://www.kamailio.org/docs/modules/devel/modules/textops.html#textops.f.… -----> This function can be used from ANY_ROUTE. OS: Debian 8 Kamailio: 4.4.5 (installed from deb repo) # kamailio -v version: kamailio 4.4.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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 4.9.2 Any suggestions? Thanks Joel.

2 4

Re: [SR-Users] Touching Base / Question
by Maxim Sobolev 17 Oct '17

17 Oct '17

len = snprintf(cmd->buf_t, sizeof(cmd->buf_t), "%d %lu %lu %lu %lu\n", ttl, pst[0].npkts_in, pst[1].npkts_in, pcnts.nrelayed, pcnts.ndropped) Those are: remaining session TTL in seconds, number of packets received from side A, number of packets received from side B, overall number of packets relayed in both directions (A->B, B->A) for directions, and number of packets dropped due to remote address not being known yet. There is actually a newer version of that command in the RTPproxy, which provides access to these and number of other per-session statistics collected by the analysis code, such as: rtpa_nsent rtpa_nrcvd rtpa_ndups rtpa_nlost rtpa_perrs The command to fetch those is for example: Q[v] %%CALLID%% %%TOTAG%% %%FROMTAG%% rtpa_nsent rtpa_nrcvd rtpa_ndups rtpa_nlost rtpa_perrs Full list is here: https://github.com/sippy/rtpproxy/blob/master/src/rtpp_command_query.c It would be good to get those implemented in the Kamailio as well. I would also probably hook up jitter values into that list. -Max On Mon, Oct 2, 2017 at 10:09 AM, Michael Mavroudis <michael(a)voicetel.com> wrote: > I'm sorry I should had added those details. I'm referring to: > > https://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html > > <https://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html> > I found a snippet at: > > http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=c > ommitdiff;h=85ee107929a80b54d47eaabef6f25d7b2f0659e7 > > There is 5 values but I'm not sure what they represent. I've also asked > Daniel so you may be hearing for him as well. > > On Sep 29, 2017 3:51 PM, "Maxim Sobolev" <sobomax(a)sippysoft.com> wrote: > >> Hi, Michael, I am not sure about October, have some other travel plans >> that are not very compatible with being in Orlando. With regards to the rtpstat, >> which counters are you using specifically? >> >> -Max >> >> On Thu, Sep 28, 2017 at 12:42 PM, Michael Mavroudis <michael(a)voicetel.com >> > wrote: >> >>> Hi, >>> >>> Hope all is well. Are you going to be in Orlando for Astricon? If not >>> I'm guessing Europe again. >>> >>> I've got a quick question for you? >>> >>> What are the contents of the psuedo var: $rtpstat >>> >>> I started calculating MOS scores and passing them to freeswitch but my >>> calculation isn't quite right. I'm thinking it's because I'm calculating >>> based on the wrong vars. Any help is greatly appreciated. >>> -- >>> Michael J. Mavroudis >>> >>> [image: VoiceTel] <http://goo.gl/yXuqFz> >>> >>> 630 Fifth Avenue >>> <https://maps.google.com/?q=630+Fifth+Avenue+New+York,+NY+10111&entry=gmail&…> >>> New York, NY 10111 >>> <https://maps.google.com/?q=630+Fifth+Avenue+New+York,+NY+10111&entry=gmail&…> >>> >>> M: (917) 426-9000 >>> W: (212) 942-7400 >>> >> >> >> >> -- >> Maksym Sobolyev >> Sippy Software, Inc. >> Internet Telephony (VoIP) Experts >> Tel (Canada): +1-778-783-0474 <%28778%29%20783-0474> >> Tel (Toll-Free): +1-855-747-7779 <%28855%29%20747-7779> >> Fax: +1-866-857-6942 <%28866%29%20857-6942> >> Web: http://www.sippysoft.com >> MSN: sales(a)sippysoft.com >> Skype: SippySoft >> > -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts Tel (Canada): +1-778-783-0474 Tel (Toll-Free): +1-855-747-7779 Fax: +1-866-857-6942 Web: http://www.sippysoft.com MSN: sales(a)sippysoft.com Skype: SippySoft

1 0

Development for Kamailio v5.1 series is frozen
by Daniel-Constantin Mierla 17 Oct '17

17 Oct '17

Hello, short note to mark the freezing of development for Kamailio v5.1 series. For few weeks, please do not push new features in the master branch. Once the branch 5.1 is created (expected to happen in 3-4 weeks from now), the master branch becomes again open for new feature. Meanwhile the focus has to be on testing current code. Work on related tools (e.g., kamctl) or documentation can still be done as well as getting the new modules in 5.1 in good shape, plus adding exports to kemi interface (which should not interfere with old code). The entire testing phase is expected to be 4 to 6 weeks, then the release of v5.1.0 -- likely by end of November should be out. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Roadmap to Kamailio v5.1
by Daniel-Constantin Mierla 16 Oct '17

16 Oct '17

Hello, being discussed at the previous IRC devel meeting before the summer, it's time to plan the preparation for next major release, respectively Kamailio 5.1 series. As an initial proposal, a timeline can be: - Mon, Oct 16, 2017 - freeze the development - in 3-4 weeks after that, create the branch 5.1 - 2-3 weeks after, do the release of v5.1.0 That means more or less by end of November we will have first version of 5.1 series. There are still 4 weeks of development, hopefully enough to get ongoing work finished. I know from github tracker that there are people working on redis support for usrloc. I will focus on exporting more functions to kemi, right now should be around 60%, by freeze should be more than 80% (likely that means all of what most of people use these days, because there are some modules that I guess are not much used). Maybe one or two modules will be pushed as well from my side. I think we should also do an IRC devel meeting to try to sync better on this. Maybe by mid of next week, I will send another email shortly. Should anyone have more details to add or adjust the proposal, do not hesitate to reply to mailing list! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 2

Having kamailio bind to a VRF device
by George Diamantopoulos 15 Oct '17

15 Oct '17

Hello, I have a use case where I need to have kamailio bind to a VRF device. The configuration in question is similar to the example below, where eth1 is a slave to the VRF-lite device: +----------+ +-------------------+ | eth0 | | vrf-green | | 1.1.1.1 | | 127.0.0.1 | +----------+ +-------------------+ | +----------+ | eth1 | | 2.2.2.2 | +----------+ Both the main routing table and "vrf-green" routing table have a default route. What I need to be able to do is have kamailio bind to both interfaces: listen=eth0:5060 listen=vrf-green:5060 And additionally be able to use force_send_socket to select an interface, for example: force_send_socket(udp:2.2.2.2:5060); However, I can't get this to work. The above configuration fails because there is no listen directive for 2.2.2.2. Also, kamailio doesn't process packets received on the VRF with the above listen directives, it behaves as if it doesn't listen on 2.2.2.2 indeed. In addition using either of the below: listen=udp:2.2.2.2:5060 or listen=eth1:5060 fails with an error upon starting kamailio. According to the kernel documentation: Applications that are to work within a VRF need to bind their socket to the VRF device: setsockopt(sd, SOL_SOCKET, SO_BINDTODEVICE, dev, strlen(dev)+1); or to specify the output device using cmsg and IP_PKTINFO. The question is, is VRF useable with kamailio right now? Or is development needed? Thanks! BR, George

5 15

rtcp-mux policy for RTPengine module
by Yuriy Gorlichenko 14 Oct '17

14 Oct '17

Hi all! Some time ago Chromium browser sets rtcpMuxPolicy: required by default (soon on Chrome 58) It means that webRTC based clients not accepts a=rtcp:31757 And uses for RTP and RTCP multiplexing at one port Main trouble that i found: calls between original SIP client and webRTC client (SIP client is initiator of call) When sip client sends invite it has a=rtcp:33445 Means it wants 2 different prots for RTCP and RTP As expected for this case webRTC client says 488 Not accessible here instead of 200 resonse I suppose rtpengine module should hept to handle it but i can not find any key how to do it I added form rtpengine_manage() rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux" But not removes a=rtcp and ice cadidate prepeared for it. Suppose removing a=rtcp:12345 will gives just an issue for RTP session. Does rtpengine module have some keys for resole this issue?

2 3

It does not log any route. Clean installation
by Bruno Henrique Gomes Gouvêa 14 Oct '17

14 Oct '17

Hello, I am doing some tests with version 4.2.3 (installed of RPM centos 7 packages) in a clean installation, I have a kamailio running in production already configured and working, but I would like to start a totally clean configuration to learn more about kamailio. I am using the configuration file that comes in the default installation (kamailio-basic.cfg) just by adding a log on the main route. I kept the log in messages file and increased the verbosity as below, and when i send any package (INVITE or REGISTER) I assume that the log or xlog lines should be displayed in file, but no. I see the packets arriving via tcpdump, and netstat shows that port 5060 is being listened by kamailio, but I have nothing else besides the initialization information (and the debug of MYSQL querys when activated unrelated to INVITE or REGISTRY packets). Am I doing something wrong? *Configs **Log :* ### LOG Levels: 3 = DBG, 2 = INFO, 1 = NOTICE, 0 = WARN, -1 = ERR #! ifdef WITH_DEBUG debug = 4 log_stderror = no #! else debug = 4 log_stderror = no #! endif memdbg = 5 memlog = 5 log_facility = LOG_LOCAL0 *Main route log test:* request_route { log ("Log main request TEST"); xlog ("Log main request TEST 2"); # not displayed route (REQINIT); */var/log/messages:* Oct 5 23:39:12 pabxmodelo kamailio: Stopping kamailio: [ OK ] Oct 5 23:39:12 pabxmodelo systemd: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Oct 5 23:39:13 pabxmodelo kamailio: INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo kamailio: Starting kamailio: Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30070]: INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Oct 5 23:39:13 pabxmodelo systemd: Started SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Oct 5 23:39:13 pabxmodelo kamailio: [ OK ] -------------------------------------------- Bruno H. G. Gouvêa ---

2 4

Re: [SR-Users] DIAMETER over SCTP for Rx and Cx Interfaces
by Carsten Bock 13 Oct '17

13 Oct '17

Hi, I am working on an update, to support SCTP for the CDP module. At the moment, I have an incomplete prototype, which binds to the Kamailio sockets (so any protocol Kamailio supports) to send/receive Diameter messages. I plan to finish it after my holiday, so I might have a full working prototype for TADsummit in Lisbon mid November. Thanks, Carsten Am 13.10.2017 14:00 schrieb "Ron McLeod" <ron.kamailio(a)mcleodnet.com>: I would like to connect Kamailio to a PCRF (Rx interface) and HSS (Cx interface) using DIAMETER over SCTP. Is this currently supported? If it is, are there any notes on how to configure this? If not, and there are plans to develop this, I would like to get involved in testing. Thanks. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

Handle SDP ICE candidates
by Thiago Guimarães 13 Oct '17

13 Oct '17

Hello, I would like to know if there is any specific configuration in kamailio.conf to enable/disable handle the array of ICE candidates. In my tests, using PJSIP for the development of my User Agent, I notice that when ICE is enabled, several calls made do not reach the other User Agent and in this case they receive a 408 error. I am suspicious that Kamailio is getting lost, when the SDP arrives with the array of ICE candidates. The initial goal of using ICE is to promote peer-to-peer communication when both are under the same router, because the application is always with STUN enabled. Best regards Thiago Barcelos Guimarães thiago.barcelos(a)gmail.com

3 4

DIAMETER over SCTP for Rx and Cx Interfaces
by Ron McLeod 13 Oct '17

13 Oct '17

I would like to connect Kamailio to a PCRF (Rx interface) and HSS (Cx interface) using DIAMETER over SCTP. Is this currently supported? If it is, are there any notes on how to configure this? If not, and there are plans to develop this, I would like to get involved in testing. Thanks.

1 0

custom kamailio-x.deb package
by Abdul Basit 13 Oct '17

13 Oct '17

Hi All, I am trying to compile custom .deb package from kamailio source that will install dependencies like mysql-server, lua, rtpproxy, etc. and will install few custom scripts along with additional db tables from tables-create.sql files. I am able to create kamailio_x-amd64.deb package that I installed on fresh machine dpkg -i kamailio_x-amd64.deb This package installed kamailio config files and modules in appropriate locations but it didn't created default, init.d and kamailio.service files. Nor it asked for installing the dependencies which I mentioned in control file. OS: Debian GNU/Linux 8.9 (jessie) Kamailio: 5.0.3 through git clone I am creating .deb package for the very 1st time. Can anyone point me to the right direction? -- regards, abdul basit

3 5

How to troubleshoot "ERROR: error while initializing modules"
by ron.kamailio＠mcleodnet.com 13 Oct '17

13 Oct '17

I have built kamailio-5.0.3 from source with no issues. I have used the P-CSCF configuration examples from ./misc/examples/pcscf and the database definitions from ./utils/kamctl/mysql. When I check the configuration file, it is reported to be "ok": # /usr/local/sbin/kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(10835) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(10835) INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* config file ok, exiting... But when I attempt to run the application, it fails with an error message "ERROR: error while initializing modules": # /usr/local/sbin/kamailio -D -d loading modules under config path: /usr/local/lib64/kamailio/modules/ Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* WARNING: no fork mode ERROR: error while initializing modules How can I troubleshoot this to determine what the actual error is? I have tried increasing the debug level, but I don't see any additional information. Thanks.

2 1

Re: [SR-Users] How to troubleshoot "ERROR: error while initializing modules" - RESOLVED
by ron.kamailio＠mcleodnet.com 12 Oct '17

12 Oct '17

More information was available in /var/log/messages: Oct 12 15:23:25 p-cscf kamailio: INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 Oct 12 15:23:25 p-cscf kamailio: ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Oct 12 15:23:25 p-cscf kamailio: ERROR: ctl [ctl.c:290]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Oct 12 15:23:25 p-cscf kamailio: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) I resolved the problem by providing the proper path for these two configuration items: modparam("ctl", "binrpc", "unix:/var/run/kamailio_pcscf/kamailio_ctl") modparam("dispatcher", "list_file", "/etc/kamailio_pcscf/dispatcher.list") -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of ron.kamailio(a)mcleodnet.com Sent: Thursday, October 12, 2017 1:36 PM To: 'Kamailio (SER) - Users Mailing List' Subject: [SR-Users] How to troubleshoot "ERROR: error while initializing modules" I have built kamailio-5.0.3 from source with no issues. I have used the P-CSCF configuration examples from ./misc/examples/pcscf and the database definitions from ./utils/kamctl/mysql. When I check the configuration file, it is reported to be "ok": # /usr/local/sbin/kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(10835) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(10835) INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* config file ok, exiting... But when I attempt to run the application, it fails with an error message "ERROR: error while initializing modules": # /usr/local/sbin/kamailio -D -d loading modules under config path: /usr/local/lib64/kamailio/modules/ Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* WARNING: no fork mode ERROR: error while initializing modules How can I troubleshoot this to determine what the actual error is? I have tried increasing the debug level, but I don't see any additional information. Thanks. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

presentce handly_subscribe() protocol/port mismatch to WS endpoint
by Yuriy Gorlichenko 12 Oct '17

12 Oct '17

Hi. I have presence server and it works fine for UDP/TCP/TLS endpoints. For now i have new one type of endpoints that runs via WebSockets It sends SUBSCRIBE request to the and then after handle_subscribe() NOTIFY not comes to the subscriber because of [core/forward.c:231]: get_send_socket2(): protocol/port mismatch I already had some issues regarding this for ACK for example but i resolved it cimply doing $ru = $ru+";transport=wss" but NOTIFY sending is internal process and can't be controlled by config file. So i can not change $ru for NOTIFY directly. Any ideas how to fix this?

2 18

Inserting local path into shared USRLOC table.
by Marrold 12 Oct '17

12 Oct '17

Hi All, I am wondering if the following is possible. For simplicity imagine two asterisk boxes and two kamailio proxies. Kamailio on the edge, asterisk in the 'core' Kamailio <=> Asterisk Asterisk <=> Kamailio A user may register to either kamailio proxy, and a call may originate from either Asterisk box where it will be sent to its *local *Kamailio proxy. >From there I'd like Kamailio to forward the request to the AoR, following the same path the registration took if it registered to the other proxy, to avoid NAT issues. I think this requires a shared USRLOC table, and also for the proxy that handled the registration to insert itself in the path of the USRLOC table. The registrar module has the 'path_check_local' parameter which sounds relevant, ignoring the path if the first hop is local. I've included add_path_received() in my REGISTRAR routing block, however I see 'Path: [not set]' in the USRLOC table. Is it possible to force the insertion of the local proxy into the path field? One additional question, in the case of NAT keepalives, is it possible to only send these from the proxy the received the register request? Thanks in advance, Matthew

4 4

kamailio QOS questions
by Lv, Victor (NSB - CN/Qingdao) 12 Oct '17

12 Oct '17

Hi All, Currently, we would like to have a try on the Kamailio QOS module toward PCRF Rx interface, it's looks a little complex to configure. Is there anyone had already successfully configured the Rx interface? If yes, could you share your configuration with me? Thanks in advance. Victor

3 3

Forcing a TCP Connection Closed?
by Mark Boyce 11 Oct '17

11 Oct '17

Hi all Just working on some connections security filters on a Kamailio install. The security goes something like this; In REQINT … if source_ip is not in customers IP white-list then just exit This works fine for UDP where packets are just ignored if they don’t come from a trusted IP. However on TCP this leads to the connection staying open until it either times out or the source disconnects. Which feels untidy. Is there a way to say close the TCP connection from within the config script? Thanks Mark -- Mark Boyce

3 5

Route Header - Comma separated issue
by Soltanici Ilie 11 Oct '17

11 Oct '17

Hi all, Why Kamailio are not removing full Route header in case if he is receiving comma separated Route? For example: Receiving such Message: SUBSCRIBE sip:101@192.168.1.1 SIP/2.0 Route: <sip:192.168.1.1;r2=on;lr=on;ftag=fb0076047b3da3e4;nat=yes>, <sip:172.16.30.10;sbc=topoh-NFlx.VXxPVXv8rfM83X1KnhF83LINVLO8VcJ6VL79zkEYH5U83hdKVhyKFi-5zX-5njs67kyYSllNx**> Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 after applying loose_route function, kamailio is forwarding this message: SUBSCRIBE sip:101@192.168.1.1 SIP/2.0 Route: Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 Other devices are sending 2 Route Headers in separate rows - and everything is working properly. Is this a bug or it should be like this per RFC? Thank You. using Kamailio 5.0.3.

1 0

[4.2.8] SUBSCRIBE message was not handled
by Jack Wang 11 Oct '17

11 Oct '17

Hello everyone, According to the routing flow set in kamailio.cfg # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); After I traced the flow it seems that SUBSCRIBE message failed on t_check_trans() and stopped there. I add some logs to keep tracing this function and found that: int t_check_trans(struct sip_msg* msg) { struct cell* t; int branch; int ret; /* already processing a T */ if(is_route_type(FAILURE_ROUTE) || is_route_type(BRANCH_ROUTE) || is_route_type(BRANCH_FAILURE_ROUTE) || is_route_type(TM_ONREPLY_ROUTE)) { return 1; } if (msg->first_line.type==SIP_REPLY) { branch = 0; ret = (t_check_msg( msg , &branch)==1) ? 1 : -1; tm_ctx_set_branch_index(branch); return ret; } else if (msg->REQ_METHOD==METHOD_CANCEL) { return w_t_lookup_cancel(msg, 0, 0); } else { switch(t_check_msg(msg, 0)){ case -2: /* possible e2e ack */ return 1; case 1: /* found */ t=get_t(); if (msg->REQ_METHOD==METHOD_ACK){ /* ack to neg. reply or ack to local trans. => process it and end the script */ /* FIXME: there's no way to distinguish here between acks to local trans. and neg. acks */ if (unlikely(has_tran_tmcbs(t, TMCB_ACK_NEG_IN))) run_trans_callbacks(TMCB_ACK_NEG_IN, t, msg, 0, msg->REQ_METHOD); t_release_transaction(t); } else { /* is a retransmission */ if (unlikely(has_tran_tmcbs(t, TMCB_REQ_RETR_IN))) run_trans_callbacks(TMCB_REQ_RETR_IN, t, msg, 0, msg->REQ_METHOD); t_retransmit_reply(t); } /* no need for UNREF(t); set_t(0) - the end-of-script t_unref callback will take care of them */ return 0; /* exit from the script */ <---------------------------- THE POINT !! } /* not found or error */ } return -1; } If the line "return 0; /* exit from the script */" was changed to "return 1; /* exit from the script */" , it works ---- means that the configuration script can keep being proceeded now. Any suggestions?

2 4

200 OK - need to include Expires
by Subba Krishna MG 11 Oct '17

11 Oct '17

Hello, This might be very simple question for you folks, I am using Kamailio as SIP - Presence server, and For every user Kamailio is sending 200 OK, but the EXPIRES IE is missing. Can u help me include this IE or any other IE for 200 OK message? How to modify the configuration files to get this done? Thanks a lot, MGSK

2 4

Path module adds either received or outbound
by Rick Barenthin 11 Oct '17

11 Oct '17

Hey, Maybe someone from the list can share some information about the path module. Why can the path module only add either the received parameter to path or the flow token with the ob parameter, but not both. Does the RFC mandates it? I didn't find anything like that and was wondering because the source calculates max length of the parameters with following: >> /* maximum possible length of suffix */ >> suffix_len = strlen(";lr;received=sip::12345%3Btransport%3Dsctp;ob;>\r\n") + IP_ADDR_MAX_STR_SIZE + 2 + (add_params ? add_params->len : 0) + 1; This looks like it can be possible to have both. Best regards Rick

2 1

how to use SIP header Route
by anil kumar k 11 Oct '17

11 Oct '17

HI, How to use SIP header Route in [SIPOUT] route instead of uri -- Regards, K.Anil Kumar

2 1

Route Not accessable
by anil kumar k 11 Oct '17

11 Oct '17

Dear Kamailio Uers, IN my [SIPOUT] by default not getting the Route from header . Any bode can help on this -- Regards, K.Anil Kumar

2 1

tsilo ts_store() causes segfault if method is not INVITE
by Anthony Alba 10 Oct '17

10 Oct '17

I was modifying the routing blocks of the tsilo push notifications recipe from Federico Cabiddu and could cause a segfault in kamailio 5.0.3 if due to misconfiguration an ACK reached ts_store(). ## modify https://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu… ## various route blocks will lead to here ## including ACK and INVITE... route[LOCALRELAY) { ## ooops: did not check method is INVITE t_relay(); ts_store(); $sht(vtp=>stored::$rU) = 1; xlog("L_INFO", "stored transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); ## saw an ACK and caused segfault } If properly checked for INVITE: if (is_method("INVITE")) { /* do tsilo stuff */ } it works. I'm wondering whether ts_store() should/could be more defensive and show an error like "..not the right context for ts_store().." . Thanks for you thoughts. Anthony

2 2

kamailio with NAT IP and DNS loadbalance
by mlwu＠south-star.com.cn 10 Oct '17

10 Oct '17

Hi, Here are same problems in my configuration of kamailio server. I have two kamailio server (both have a NAT IP and a lan IP), and I use DNS to keep loadbalance.When sending message to another account , if NAT IP which is returned by DNS, is not the same as receiver account, kamailio server sends 404 not found error. I really have no idea how to set the configuration when using DNS loadbalance to return two NAT IP. Thank you , I'm looking forward from you! Sincerely yours Ada WU mlwu(a)south-star.com.cn

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