- sr-users - kamailio.org

Regd. Stored procedure support in Kamailio
by Shankar 04 Mar '14

04 Mar '14

Hello All, Does Kamailio support invoking stored procedures instead of sending plain DB queries already? We have been asked to migrate to stored procedures. I am trying to estimate the effort involved in migrating Kamailio code to support stored procedure. Other question that comes to my mind is whether stored procedures are really needed in Kamailio. Can someone please clarify? Regards, Shankar

1 0

REGISTER failure_route 401 problem
by Marc Soda 03 Mar '14

03 Mar '14

I have a Kamailio server setup which is registers to a back end server on behalf of endpoints. The endpoints can register to Kamailio but Kamailio is failing to register to the server when I put a NAT device in front of it. Without the NAT device it works fine. The problem is the 401 that comes back seems to be ignored by Kamailio. I have a failure route setup to auth, but it is never hit. I see the 401 in onrely_route, but not the failure_route. I'm assuming it's a NAT issue because removing the device fixes the issue. Anyone have any ideas? The 401 being ignored: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.10.11;branch=z9hG4bKe5d6.178378f7.0;received=198.XXX.XXX.XXX Via: SIP/2.0/UDP 127.0.0.1:12354 ;rport=6545;received=198.XXX.YYY.YYY;branch=z9hG4bK-1879-1-3 From: <sip:sip7878_spqa@64.YYY.YYY.YYY>;tag=1 To: <sip:sip7878_spqa@64.YYY.YYY.YYY>;tag=as00e32130 Call-ID: 1-1879(a)127.0.0.1 CSeq: 2 REGISTER User-Agent: CoreDialPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="fe-c7c5-9o.domain.com", nonce="151e4f60" Content-Length: 0 Thanks, Marc

2 6

usrloc preload
by Victor V. Kustov 03 Mar '14

03 Mar '14

Hi all! Please give expanded tips about usrloc preload(). What exactly it doing? modparam("usrloc", "db_mode", 0) modparam("usrloc", "preload", "location") This case kamailio load from db on startup and operate in memory after? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.12.9-calculate GNU/Linux

3 5

Ubuntu precise apt repositories unavailable
by Josef Stautner 03 Mar '14

03 Mar '14

Goor morning community, I received a notification that the repositories for Ubuntu Precise are not available anymore. /W: Failed to fetch //http://deb.kamailio.org/kamailio/dists/precise/Release.gpg// Could// //not connect to deb.kamailio.org:80 (87.106.140.63). - connect (110: Connection timed out)// / / //W: Failed to fetch //http://deb.kamailio.org/kamailio/dists/precise/main/source/Sources//// //Unable to connect to deb.kamailio.org:http:// / / //W: Failed to fetch// / /http://deb.kamailio.org/kamailio/dists/precise/main/binary-i386/Packages// Unable to// //connect to deb.kamailio.org:http:// / / //W: Failed to fetch// / /http://deb.kamailio.org/kamailio/dists/precise/main/i18n/Translation-en_US// Unable to// //connect to deb.kamailio.org:http:// / / //W: Failed to fetch// / /http://deb.kamailio.org/kamailio/dists/precise/main/i18n/Translation-en// Unable to// //connect to deb.kamailio.org:http:// / After I tried manually to fetch new updates I got a connection and downloaded some data but then the connection timed out. Is this just a temporary probleme and are there any fallback mirrors? Thanks in advance Josef Stautner

2 2

Adding Path header for the current request.
by MAXX ... 02 Mar '14

02 Mar '14

I found the solution myself, anyone having a similar issue in the future can use msg_apply_changes() from textopsx module after append_hf line.

1 0

Adding Path header for the current request.
by MAXX ... 02 Mar '14

02 Mar '14

Hello, I need to add a Path header to the current request so the save method of registrar module will save the location with the path value to the db. I made the necessary settings for the registrar module but I couldn't be able to add the header. Neither "append_hf" nor "add_path" of path module works since they don't add the header to the currently processing request. $hdr pseudovariable is readonly so I couldn't modify it as well. What can I do to make this happen? Also the reason I need this is to let people registered to different registrars to be able to call each other via their current tls connections to their respective registrars. Thanks In Advance.

1 0

XAVP in 3.1.6
by Jijo 01 Mar '14

01 Mar '14

Hi I'm trying to use XAVP but the script is showing parsing error. Do i need to include any module? I rebuild kamailio using the flag WITH_XAVP but didn' help. route[0] { $xavp(a=>bar) = 'bar'; } S-154:~ # /usr/sbin/kamailio -c -f /etc/kamailio/kam86x2.cfg loading modules under /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(20184) ERROR: <core> [pvapi.c:516]: error searching pvar "xavp" 0(20184) ERROR: <core> [pvapi.c:720]: wrong char [a/97] in [$xavp(a=>bar)] at [6 (5)] 0(20184) : <core> [cfg.y:3461]: parse error in config file /etc/kamailio/kam86x2.cfg, line 555, column 5-17: unknown script pseudo variable $xavp(a=>bar) ERROR: bad config file (1 errors) Segmentation fault (core dumped) S-154:~ # /usr/sbin/kamailio -V version: kamailio 3.1.6 (i386/linux) af51bc flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 25MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: af51bc compiled on 14:11:43 Feb 28 2014 with gcc 4.7.1 THANKS

1 0

How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?
by Alex Villacís Lasso 01 Mar '14

01 Mar '14

As part of a project, I have installed a CentOS 6 test system (a virtual machine) with Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_…. I am trying to setup a combination of Kamailio and Asterisk that will route SIP calls between all the configured networks in the test setup, in addition to being capable of using Asterisk in order to handle PSTN and IAX2 calls. I am using the following online guide to modify my kamailio.cfg: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . Based on this, I generated the attached patch for my Kamailio configuration My test setup has the following network interfaces: eth0: 10.1.0.3, on network 10.1.0.0/24 eth1: 192.168.5.18, on network 192.168.0.0/16 eth2: 10.0.0.2, on network 10.0.0.0/24 lo: 127.0.0.1, on network 127.0.0.0/8 I first configured Asterisk with SIP realtime support (with no Kamailio), and tested that all configured accounts could register from all interfaces, and that Asterisk could properly route media between any two disjoint networks. After installing Kamailio, the guide called for disabling Asterisk SIP authentication by setting passwords to NULL, and moving Asterisk SIP to a different port (I chose 5080) so that Asterisk and Kamailio could run on the same machine. At this point, the SIP clients (one softphone and one VoIP phone) can now register at port 5080 without authentication. In the process of changing my Kamailio configuration according to the attached patch, the guide says that I should configure the IP of the network interface as the value of asterisk.bindip and kamailio.bindip. After performing all required changes, Kamailio does take over authentication at the default port of 5060. Testing shows that for all SIP clients with IPs belonging to the same network as the configured asterisk.bindip, both registration and media exchange work correctly, and that the SIP clients are still capable of calling into the Asterisk dialplan, and therefore, routing into Asterisk resources. For SIP clients in disjoint networks, the failure mode depends on whether mhomed is enabled or disabled in kamailio.cfg. For mhomed=0 (or unset), I have the following situation between the two SIP clients (one at 10.1.0.1, the other at 10.0.0.3), as shown by "sip show peers" in Asterisk (when asterisk.bindip is set to 192.168.5.18): Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. Name/username Host Dyn Forcerport ACL Port Status Description Realtime gatitoscomx64am_100/gatit 10.1.0.3 D N A 5060 OK (16 ms) Cached RT gatitoscomx64am_101/gatit 10.0.0.2 D N A 5060 OK (36 ms) Cached RT gatitoscomx64am_IM101 (Unspecified) D N A 0 UNREACHABLE Cached RT 3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline] If I try to call from one SIP client to an extension in the Asterisk dialplan that does NOT map to a SIP client in a disjoint network, the media exchange works (with negotiatied media IP in the same network as the SIP client), regardless of whether the calling client belongs in the same network as asterisk.bindip. If I try to call from the same SIP client to an extension that maps to a SIP client in a disjoint network, the call fails, and I get the spoken message about the user at extension such-and-such being unavailable. Additionally, I get the following error message in the Asterisk logs: [Feb 25 16:53:14] NOTICE[13807][C-00000003] chan_sip.c: Call from 'gatitoscomx64am_101' (10.0.0.2:5060) to extension 'gatitoscomx64am_101' rejected because extension not found in context 'gatitoscomx64am-from-internal'. For mhomed=1, the output of "sip show peers" changes to the following (when asterisk.bindip is set to 192.168.5.18): Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. Name/username Host Dyn Forcerport ACL Port Status Description Realtime gatitoscomx64am_100/gatit 192.168.5.18 D N A 5060 OK (19 ms) Cached RT gatitoscomx64am_101/gatit 192.168.5.18 D N A 5060 OK (34 ms) Cached RT gatitoscomx64am_IM101 (Unspecified) D N A 0 UNREACHABLE Cached RT 3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline] From wireshark sniffing, I can see that the SDP payload sent from the client to Kamailio contains the IP address of the client, which is accessible by both Kamailio and Asterisk. However, the SDP payload in the OK response sent back to the client contains a media port with the IP address of asterisk.bindip (the one that appears in the "Host" column in the "sip show peers" report), not the IP address of the interface that received the INVITE. This results in broken media negotiation for all SIP clients belonging to networks other than the one that contains asterisk.bindip. In either case, I have to hardcode an IP address in kamailio.cfg, which is not satisfactory. IPs assigned to interfaces can and do change, especially if the interface is managed with DHCP. To escape this, I tried setting asterisk.bindip to 127.0.0.1, but since apparently localhost is also a disjoint network, all of the above described problems apply. Related to these issues, I am not satisfied with leaving Asterisk running unauthenticated SIP at the nonstandard port. Somebody suggested blocking the port with iptables, but I do not want to rely on this alone. I tried setting bindaddr=127.0.0.1 so that only Kamailio gets to talk to Asterisk, but this also has the side effect of restricting the media negotiation to localhost only. I am asking for help in building a Kamailio/Asterisk configuration that will support all of the networks and route media between all of them, just as if Asterisk were the only program running. Ideally, the configuration should not encode the current IP of any interface (except, maybe, localhost). What is the official name (if any) for the setup I am describing above? Does it have a standard setup procedure? How is Asterisk secured so that clients cannot bypass authentication using the Asterisk SIP port directly?

2 4

Replacing an ACME Packet Net-Net SBC
by Francesco Maria Magnini 01 Mar '14

01 Mar '14

Hi, I would like to have some suggestions about a full replacement of an ACME Packet Net-Net Session Border Controller. By now, ACME SBC performs all the SBC functionalities, mainly: - it is used as a SIP endpoint for SIP client registrations - it is used as a SIP endpoint for interconnection to multiple SIP carriers via SIP trunks - it is used for NAT traversal In this deployment, the SIP Server communicates only with the SBC and this one takes care of the communication between the SIP Server and the external SIP entities (UA clients, SIP Trunks). In this scenario, can I consider to replace the SBC with Kamailio?

13 22

prevent relaying to myself
by Henry Fernandes 01 Mar '14

01 Mar '14

Is there a way to prevent relaying requests to myself in route[RELAY]? -H

3 3

fresh kamailio installation unable to connect through jitsi
by Michelle Jun 01 Mar '14

01 Mar '14

hi i just finished installation of kamailio on centos 6 64 bit following the docs on http://www.fredposner.com/voip/1457/kamailio-behind-nat/ the user able to chat through jitsi just fine, but when calling, one user able to connect, while the other showing only "connecting" (audio/video) any idea what did i do wrong? here is the /var/log./messages thank you Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session 2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0, tag 2d9e2107;1 requested, type strong Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session on a port 25982 created, tag 2d9e2107;1 Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: pre-filling caller's address with 192.168.168.1:5000 Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session 2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0, tag 2d9e2107;2 requested, type strong Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session on a port 25192 created, tag 2d9e2107;2 Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: pre-filling caller's address with 192.168.168.1:5002 Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: pre-filling callee's address with 192.168.168.1:5000 Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: pre-filling callee's address with 192.168.168.1:5002 Feb 26 17:43:39 black96 /usr/local/sbin/kamailio[4456]: NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1393465419;method=INVITE;from_tag=2d9e2107;to_tag=7175e4ad;call_id=2f6e045de764ae9e9252352d952359ae@0 :0:0:0:0:0:0:0;code=200;reason=OK;src_user=handy;src_domain=184.105.148.231;src_ip=192.168.168.1;dst_ouser=harri;dst_user=harri;dst_domain=192.168.113.2 Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:43:40 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:40 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:43:42 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:42 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:process_rtp: session timeout Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: session on ports 25982/27044 is cleaned up Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:process_rtp: session timeout Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: session on ports 25192/22432 is cleaned up Feb 26 17:45:23 black96 rtpproxy[4293]: INFO:handle_command: delete request failed: session 2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0, tags 2d9e2107/7175e4ad not found Feb 26 17:45:53 black96 /usr/local/sbin/kamailio[4460]: NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1393465553;method=BYE;from_tag=2d9e2107;to_tag=7175e4ad;call_id=2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0;code=408;reason=Request Timeout;src_user=handy;src_domain=184.105.148.231;src_ip=192.168.168.1;dst_ouser=harri;dst_user=;dst_domain=184.105.148.231

3 7

What is required to enable TLS on non-standard port?
by info＠vintageelectronics.ca 01 Mar '14

01 Mar '14

Kamailio running in Fedora 19 box. The client (Jitsi) can only connect over TCP. If connecting over UDP, the clients connect, but cannot send/receive messages. Cannot connect over TLS at all. Ultimately I will need to run with TLS over 5065 port or higher. Tried using different listen directives, different port directive, but kamailio either does not start, or only listening on one TCP port. I've set up two SRV records, one for _sip on port 5065 and another for _sips on port 5066. Both pointing to the Fedora box. In Jitsi I have to specify port # and when I set it to 5065 over TCP it works, but 5066 over TLS never connects. What am I doing wrong? Thank you.

3 8

Kamailio crashes with double qm_free
by Tuan Viet Nguyen 28 Feb '14

28 Feb '14

Hello, I have another kamailio crash with 4.1.1. The scenario is simple: + call a number, fail + retry on another peer, fails In this case, I use t_reply("404", "Not found") then exit. It seems that we have a double qm_free. Log file Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29595]: WARNING: tm [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release transaction Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29583]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f9dedb3e110), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29627]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 19 Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29576]: ALERT: <core> [main.c:775]: handle_sigs(): child process 29583 exited by a signal 6 Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29576]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated core.kamailio.29583 (gdb) bt full #0 0x00007f9df6afb475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f9df6afe6f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x0000000000548253 in qm_free (qm=0x7f9ded878000, p=0x7f9dedb3e110, file=0x7f9df515b62d "tm: h_table.c", func=0x7f9df515b7d8 "free_cell", line=157) at mem/q_malloc.c:470 f = 0x7f9dedb3e0e0 size = 140316274568736 next = 0x0 prev = 0x0 __FUNCTION__ = "qm_free" #3 0x00007f9df50ee2b3 in free_cell (dead_cell=0x7f9dedb39490) at h_table.c:157 b = 0x0 i = 2 rpl = 0x0 tt = 0x177dd596e30 foo = 0x7fffdd597400 cbs = 0x7f9dedb39490 cbs_tmp = 0x7fffdd596f10 __FUNCTION__ = "free_cell" #4 0x00007f9df511c662 in t_unref (p_msg=0x7f9df68bfdb0) at t_lookup.c:1546 kr = 12 __FUNCTION__ = "t_unref" #5 0x00007f9df5146f4d in w_t_unref (foo=0x7f9df68bfdb0, flags=2147483649, bar=0x0) at tm.c:765 No locals. #6 0x00000000004d6f27 in exec_post_script_cb (msg=0x7f9df68bfdb0, type=REQUEST_CB_TYPE) at script_cb.c:195 cb = 0x7f9df68a97d0 flags = 2147483649 #7 0x00000000004a6e24 in receive_msg (buf=0x921620 "ACK sip:0123456789@10.100.8.7 SIP/2.0\r\nVia: SIP/2.0/UDP 10.100.8.94;branch=z9hG4bKbcd8.9b0d4b2682a07fefaa22406761659624.0\r\nMax-Forwards: 15\r\nFrom: \"via_test\" <sip:0310193301@10.100.8.12>;tag=as4d03043"..., len=375, rcv_info=0x7fffdd5970d0) at receive.c:228 msg = 0x7f9df68bfdb0 ctx = {rec_lev = -581341280, run_flags = 32767, last_retcode = 5, jmp_env = {{__jmpbuf = {8857080, 0, 0, 0, 0, 140316433164786, 17179869189, 0}, __mask_was_saved = 8856992, __saved_mask = {__val = {1, 140316271892032, 1656210553, 3713626240, 1024, 8008593472, 140316271892032, 140736907014208, 5472467, 1577608202, 140316271892032, 50195, 140316271892032, 140316426284544, 8282899704, 140736907014272}}}}} ret = 0 inb = {s = 0x921620 "ACK sip:0123456789@10.100.8.7 SIP/2.0\r\nVia: SIP/2.0/UDP 10.100.8.94;branch=z9hG4bKbcd8.9b0d4b2682a07fefaa22406761659624.0\r\nMax-Forwards: 15\r\nFrom: \"via_test\" <sip:0310193301@10.100.8.12>;tag=as4d03043"..., len = 375} __FUNCTION__ = "receive_msg" #8 0x000000000053c0d8 in udp_rcv_loop () at udp_server.c:536 len = 375 buf = "ACK sip:0123456789@10.100.8.7 SIP/2.0\r\nVia: SIP/2.0/UDP 10.100.8.94;branch=z9hG4bKbcd8.9b0d4b2682a07fefaa22406761659624.0\r\nMax-Forwards: 15\r\nFrom: \"via_test\" <sip:0310193301@10.100.8.12>;tag=as4d03043"... tmp = 0x13e0751a0eb8aa4f <Address 0x13e0751a0eb8aa4f out of bounds> from = 0x7f9df68a5408 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {1577608202, 4290224}, addr32 = {1577608202, 0, 4290224, 0}, addr16 = {25610, 24072, 0, 0, 30384, 65, 0, 0}, addr = "\nd\b^\000\000\000\000\260vA\000\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = { 117990410, 0}, addr32 = {117990410, 0, 0, 0}, addr16 = {25610, 1800, 0, 0, 0, 0, 0, 0}, addr = "\nd\b\a", '\000' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023\304\nd\b^\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 1577608202}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 1577608202, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x7f9df6704da8, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #9 0x000000000046ec98 in main_loop () at main.c:1617 i = 5 pid = 0 si = 0x7f9df6704da8 si_desc = "udp receiver child=5 sock=10.100.8.7:5060\000\000\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000y\304\267b\000\000\000\000\260vA\000\000\000\000\000\000tY\335\377\177", '\000' <repeats 18 times>, "@rY\335\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #10 0x0000000000471c38 in main (argc=5, argv=0x7fffdd597408) at main.c:2533 cfg_stream = 0x164c010 c = -1 r = 0 tmp = 0x7fffdd597438 "\211~Y\335\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 4192398120 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" core.kamailio.29576 (gdb) bt full #0 0x00007f9df6afb475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f9df6afe6f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x0000000000548253 in qm_free (qm=0x7f9ded878000, p=0x7f9dedb3e110, file=0x7f9df515b62d "tm: h_table.c", func=0x7f9df515b7d8 "free_cell", line=157) at mem/q_malloc.c:470 f = 0x7f9dedb3e0e0 size = 140316274568736 next = 0x0 prev = 0x0 __FUNCTION__ = "qm_free" #3 0x00007f9df50ee2b3 in free_cell (dead_cell=0x7f9dedb39490) at h_table.c:157 b = 0x7f9df464b480 "dialog: dlg_cb.c" i = 0 rpl = 0x7f9df464b8e0 tt = 0x7f9df4a8cc75 foo = 0x5000548a29 cbs = 0xd000000001 cbs_tmp = 0x1edb086c8 __FUNCTION__ = "free_cell" #4 0x00007f9df50ef480 in free_hash_table () at h_table.c:441 p_cell = 0x7f9dedb39490 tmp_cell = 0x7f9deda23d90 i = 36299 __FUNCTION__ = "free_hash_table" #5 0x00007f9df5102d35 in tm_shutdown () at t_funcs.c:122 __FUNCTION__ = "tm_shutdown" #6 0x00000000004f8101 in destroy_modules () at sr_module.c:817 t = 0x7f9df670fd50 foo = 0x7f9df670f588 __FUNCTION__ = "destroy_modules" #7 0x00000000004689b2 in cleanup (show_status=1) at main.c:560 memlog = 32669 __FUNCTION__ = "cleanup" #8 0x0000000000469aab in shutdown_children (sig=15, show_status=1) at main.c:702 __FUNCTION__ = "shutdown_children" #9 0x000000000046b146 in handle_sigs () at main.c:793 chld = 0 chld_status = 134 memlog = 0 __FUNCTION__ = "handle_sigs" #10 0x000000000046f549 in main_loop () at main.c:1746 i = 8 pid = 29627 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000y\304\267b\000\000\000\000\260vA\000\000\000\000\000\000tY\335\377\177", '\000' <repeats 18 times>, "@rY\335\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #11 0x0000000000471c38 in main (argc=5, argv=0x7fffdd597408) at main.c:2533 cfg_stream = 0x164c010 c = -1 r = 0 tmp = 0x7fffdd597438 "\211~Y\335\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 4192398120 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" Thanks for your help

2 2

kamailio crashes
by Tuan Viet Nguyen 28 Feb '14

28 Feb '14

Hello, I have a crash with the following scenario: I try to route an INVITE to a remote host, it fails, I route to the second suing the dispatcher module. It fails the second time and I stop the call. It seems that I did not release the transaction in this case 93454 Feb 28 11:39:08 kamailio23 /usr/local/sbin/kamailio[20225]: WARNING: tm [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release transaction And then, Kamailio receives BYE of other dialogs and it crashes Here is core bt of 2 kamailio processes (4.1.1) (gdb) bt full #0 0x0000000000534e4e in timer_list_expire (t=1283469267, h=0x7ff9219e99b8, slow_l=0x7ff9219e9c88, slow_mark=12) at timer.c:883 tl = 0x7ff921c6b8f0 ret = 32767 #1 0x00000000005351ba in timer_handler () at timer.c:959 saved_ticks = 1283469267 run_slow_timer = 0 i = 12 __FUNCTION__ = "timer_handler" #2 0x0000000000535453 in timer_main () at timer.c:998 No locals. #3 0x000000000046efa2 in main_loop () at main.c:1688 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000\243Mgf\000\000\000\000\260vA\000\000\000\000\000\020\201\351a\377\177", '\000' <repeats 18 times>, "P\177\351a\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #4 0x0000000000471c38 in main (argc=5, argv=0x7fff61e98118) at main.c:2533 cfg_stream = 0x1970010 c = -1 r = 0 tmp = 0x7fff61e98148 "\211\236\351a\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 519948236 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" (gdb) bt full #0 0x00007ff92abf3475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007ff92abf66f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x0000000000548253 in qm_free (qm=0x7ff921970000, p=0x7ff921c3b940, file=0x7ff92925362d "tm: h_table.c", func=0x7ff9292537d8 "free_cell", line=178) at mem/q_malloc.c:470 f = 0x7ff921c3b910 size = 5600 next = 0x7fff61e97ac0 prev = 0x7ff9291e571e __FUNCTION__ = "qm_free" #3 0x00007ff9291e64a9 in free_cell (dead_cell=0x7ff921c6b870) at h_table.c:178 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x5000548a29 cbs = 0x0 cbs_tmp = 0x7ff921c3b940 __FUNCTION__ = "free_cell" #4 0x00007ff9291e7480 in free_hash_table () at h_table.c:441 p_cell = 0x7ff921c6b870 tmp_cell = 0x7ff921a159f0 i = 37066 __FUNCTION__ = "free_hash_table" #5 0x00007ff9291fad35 in tm_shutdown () at t_funcs.c:122 __FUNCTION__ = "tm_shutdown" #6 0x00000000004f8101 in destroy_modules () at sr_module.c:817 t = 0x7ff92a807d50 foo = 0x7ff92a807588 __FUNCTION__ = "destroy_modules" #7 0x00000000004689b2 in cleanup (show_status=1) at main.c:560 memlog = 32761 __FUNCTION__ = "cleanup" #8 0x0000000000469aab in shutdown_children (sig=15, show_status=1) at main.c:702 __FUNCTION__ = "shutdown_children" #9 0x000000000046b146 in handle_sigs () at main.c:793 chld = 0 chld_status = 139 memlog = 0 __FUNCTION__ = "handle_sigs" #10 0x000000000046f549 in main_loop () at main.c:1746 i = 8 pid = 20250 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000\243Mgf\000\000\000\000\260vA\000\000\000\000\000\020\201\351a\377\177", '\000' <repeats 18 times>, "P\177\351a\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #11 0x0000000000471c38 in main (argc=5, argv=0x7fff61e98118) at main.c:2533 cfg_stream = 0x1970010 c = -1 r = 0 tmp = 0x7fff61e98148 "\211\236\351a\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 519948236 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" Thank you for your help

2 1

Add Header
by Tony Turner 28 Feb '14

28 Feb '14

Hi, customers PBX----trunk-----kamailio------------freeswitch-------------carrier We already add Trusted IP’s via SIREMIS 4.0 is it possible with Kamailio to have customers calls come in on a trusted IP and when they send calls we add a header with a [client id] which we can remove on the next ip switch upstream So example Call comes in from customer with IP: their trusted IP source xx.xx.xx.xx add header with their client ID route to PSTN Then my upstream switch will process the customers call and remove the header before sending to our voip carrier. Basically my ip switch will evaluate the client id header for billing, process and strip the header back out. Thanks Tony

2 2

proposed patch for the registrar module - crash fix
by Dragos Oancea 28 Feb '14

28 Feb '14

Hi We think that we found and fix a bug in the registrar module. The bug is hard to reproduce, and it crashes our Kamailio from time to time (once at 2-3 weeks for us) . In save.c, function update_contacts() there are two places where we free() a pointer and then we reuse it, line 700 and line 730: while(ptr){ ptr0 = ptr; if(ptr!=c) ul.delete_ucontact(_r, ptr); ptr=ptr0->next; } And then from inside delete_ucontact(_r,ptr) we call mem_delete_ucontact(_r, _c) which calls free_ucontact(_c) , which calls shm_free(_c) . _c is actually our ptr. If another process writes at the location pointed by ptr during the "while" loop , the current process will crash. The bug affects 4.0.4, 4.0.5 and maybe older versions. We keep the location table in memory, no database backend, and we do alot of REGISTER/un-REGISTER in our environment, and maybe that's why this bug was not spotted by the community before. Just to clarify, this fix is in relation with this post: http://lists.sip-router.org/pipermail/sr-dev/2014-February/022934.html Please take into consideration the attached patch provided by the Libon Voice Team. Regards, Dragos Oancea

2 1

Mobile 3G Block from provider
by Rene Montilva 28 Feb '14

28 Feb '14

Hi List I'm trying to call over 3G networks, but doesn't work, i think my mobile provider block voip traffic, is there any way to hide voip traffic or jump the provider block rule ??

2 1

keep buddy list - best way
by Mihai Marin 27 Feb '14

27 Feb '14

Hello Sirs, I'm a little confused getting a decision about implementing buddy list (step: presence & IM). What is the best way for keeping the contacts list: xmpp or xcap? You will say probably xcap server but even if I intend to use in the future an xmpp gateway in order to be able to communicate sip with with xmpp servers like google - google talk? And why xcap? Because most of the clients has the possibility to connect to a xcap server to get the buddy list? From web perspective I think both are OK because of the xhttp module power in kamailio to get the buddy list xml. Thank you. Best regards, Mihai M

1 0

kamailio error, failed to start.
by Arya Farzan 27 Feb '14

27 Feb '14

I just installed kamailio on Debian 7 and when I start it with init.d I get the following messages. root@voip:/etc/kamailio# /etc/init.d/kamailio start Starting Kamailio: loading modules under /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: MYSERVERIP:5060 tcp: 127.0.0.1:5060 tcp: MYSERVERIP:5060 tls: 127.0.0.1:5061 tls: MYSERVERIP:5061 Aliases: tls: voip:5061 tls: localhost:5061 tcp: voip:5060 tcp: localhost:5060 udp: voip:5060 udp: localhost:5060 kamailio error, failed to start. root@voip:/etc/kamailio# How can I find out what the problem is?

3 4

Re: [SR-Users] Replacing an ACME Packet Net-Net SBC
by Raúl Alexis Betancor Santana 27 Feb '14

27 Feb '14

Sorry ... I disagree at all .... All that features you mention, could be done with kamalio ... BUT you have to work on your cfg to get them. You don't get then out of the box. Kamailio it's a powerfull tool ... It's not a final product as the AP Net SBC Best regards Luis Silva <luisfilsilva(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

kamailio with mediaproxy-ng, 488 Not Acceptable Here
by Mihai Marin 26 Feb '14

26 Feb '14

Hello Sirs, I have a problem configuring kamailio with mediaproxy-ng and I'm asking for help. I managed to build everything, kamailio find support for mediaproxy-ng using rtpproxy-ng. When I'm trying to make a call from Web using my phone's internet provider to my computer's web I get 488 Not Acceptable Here. Swithing the caller I get no video. I have used the kamailio-advanced.cfg generated and added websocket support. The call in my network is working perfect. Can you help me with this? mediaproxy-ng log: mediaproxy-ng[14896]: Returning to SIP proxy: d7:createdi1390864117e7:streamslld3:tag10:trhh9viigs6:status34:known but unconfirmed peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee10:local porti30008ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee10:local porti30009eeeed3:tag0:6:status20:unknown peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30010ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30011eeeeee6:totalsd5:inputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eee6:outputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eeee6:result2:oke kamailio: 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=860 a=24 n=rtpproxy_manage 11(14926) DEBUG: rtpproxy-ng [rtpproxy.c:1333]: rtpp_function_call(): proxy reply: d7:createdi1390864117e7:streamslld3:tag10:trhh9viigs6:status34:known but unconfirmed peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee10:local porti30008ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee10:local porti30009eeeed3:tag0:6:status20:unknown peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30010ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30011eeeeee6:totalsd5:inputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eee6:outputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eeee6:result2:oke 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=867 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=862 a=24 n=is_request 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=866 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=863 a=24 n=has_totag 11(14926) DEBUG: siputils [checks.c:103]: has_totag(): no totag 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=864 a=25 n=add_rr_param 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=873 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=867 a=24 n=is_reply 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=873 a=2 n=return 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=1009 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=989 a=24 n=t_is_canceled 11(14926) DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=0x7f62174df4e0 11(14926) DEBUG: tm [t_lookup.c:1143]: t_check_msg(): DEBUG: t_check_msg: T already found! 11(14926) DEBUG: tm [t_reply.c:1827]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 11(14926) DEBUG: <core> [msg_translator.c:2007]: generate_res_buf_from_sip_res(): old size: 397, new size: 305 11(14926) DEBUG: <core> [msg_translator.c:2025]: generate_res_buf_from_sip_res(): copied size: orig:125, new: 33, rest: 272 msg= SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS ebhg3v0qb6fm.invalid;rport=1034;received=213.233.85.55;branch=z9hG4bK8048296 To: <sip:bob@93.187.138.214>;tag=r6dc2287g9 From: "Alice Test" <sip:alice@93.187.138.214>;tag=trhh9viigs Call-ID: fvgga4ikm8vrvuji0g0n CSeq: 6228 INVITE Content-Length: 0 11(14926) DEBUG: websocket [ws_frame.c:713]: ws_frame_transmit(): Tx message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS ebhg3v0qb6fm.invalid;rport=1034;received=213.233.85.55;branch=z9hG4bK8048296 To: <sip:bob@93.187.138.214>;tag=r6dc2287g9 From: "Alice Test" <sip:alice@93.187.138.214>;tag=trhh9viigs Call-ID: fvgga4ikm8vrvuji0g0n CSeq: 6228 INVITE Content-Length: 0 Thank you. Best regards, Mihai M

8 27

coredump on using jsonprc_request
by davy van de moere 26 Feb '14

26 Feb '14

Gents, I’ve stumbled upon a coredump when using jsonrpc_request. I might be expecting the wrong things from jsonrpc_request. I was hoping to use it in combination with CNXCC, and have jsonrpc_request get a maximum duration for every call, as a fraud avoidance mechanism. I tried this with the branch 4.1 and master. Both behave identical. So from the kamailio-basic.cfg example, I just added this line around line 350 (just above route(SIPOUT)): jsonrpc_request("bar.sendsms", "{'ru':'$rU','rb':'$rb'}", "JSONRESPONSE", "JSONERROR", "$var(result)"); Doing a ngrep on the netstrings I have: interface: venet0 (127.0.0.2/255.255.255.255) filter: (ip or ip6) and ( port 7080 ) # T 10.66.8.42:44995 -> 192.168.60.45:7080 [AP] 94:{ "id": 2, "jsonrpc": "2.0", "method": "bar.sendsms", "params": { "ru": "", "rb": "" } }, ## T 192.168.60.45:7080 -> 10.66.8.42:44995 [AP] 43:{"jsonrpc": "2.0", "result": "12", "id": 2}, ####^Cexit 7 received, 0 dropped I do get some results from the JSONRESPONSE route. But when launching a call, or when sending just a register packet, kamailio gives up with: Feb 26 06:29:11 SBC01 kamailio[18043]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Feb 26 06:29:12 SBC01 kamailio[18033]: : [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb4c01624), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Feb 26 06:29:12 SBC01 kamailio[18023]: ALERT: [main.c:775]: handle_sigs(): child process 18033 exited by a signal 6 Feb 26 06:29:12 SBC01 kamailio[18023]: ALERT: [main.c:778]: handle_sigs(): core was generated Feb 26 06:29:12 SBC01 kamailio[18023]: INFO: [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Feb 26 06:29:12 SBC01 kamailio[18041]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18047]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18043]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18048]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18023]: : [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb4c01624), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Using the jsonrpc_notification works like a charm. I got coredumps of it, if someone is interested. Side questions are offcourse, am I aiming to use jsonrpc_request for the correct purpose? Are there any other good alternatives? (next to using plain db queries?) Grtz, Davy -- davy van de moere

1 0

Kamailio World 2014 - Agenda
by Daniel-Constantin Mierla 26 Feb '14

26 Feb '14

The first version of the schedule during Kamailio World 2014 <http://www.kamailioworld.com> [1] has been published. Over 30 sessions grouped in two days of conference and pre-conference technical tutorials, spanning from various use cases for Kamailio to presentations approaching related VoIP applications such as Jitsi, Sems, Asterisk or FreeSwitch. Among the topics: WebRTC, IMS or VoIP security and fraud detection, tools for testing and troubleshooting VoIP systems, scalability with OpenStack and asynchronous processing, real time charging. First version of agenda has been published at: - http://conference.kamailio.com/k02/schedule/ Information about speakers can be found at: - http://conference.kamailio.com/k02/speakers/ Many thanks to the sponsors [2], they are making possible this event: FhG Fokus <http://www.fokus.fraunhofer.de/en/fokus/index.html>, Asipto <http://www.asipto.com>, Sipwise <http://www.sipwise.com>, Sipgate <http://www.sipgate.de>, IT Center <http://www.itcenter.com.pt/>, Simwood <http://www.simwood.com>, and NG Voice <http://www.ng-voice.com/>. Should you consider sponsoring, contact us via email at <conference [at] kamailio.com>**and we can present you more details. If you plan to attend, do not delay the registration, the number of seats is limited – see more details in Registration page <http://conference.kamailio.com/k02/registration/> [3]. /Looking forward to meeting many of you in Berlin! / Daniel/ / [1] - http://conference.kamailio.com [2] - http://conference.kamailio.com/k02/sponsors/ [3] - http://conference.kamailio.com/k02/registration/ -- Daniel-Constantin Mierla http://www.asipto.com

1 0

debian packages from opensuse.org
by Jöran Vinzens 26 Feb '14

26 Feb '14

Hi All, we are using Kamailio 4.1.0 packages from opensuse: http://download.opensuse.org/repositories/home:/kamailio:/telephony-debs/De… During installation it seems, the packages are malformed. Installation of kamailio-mysql-modules and kamailio-xml-modules stops during unpacking with failure: "trying to overwrite '/usr/lib64/kamailio/libsrdb2.so.1.0', which is also in package kamailio 4.1.0" it seem this one file is part of all three packages. There is a way to enforce the installation using dpkg -i --force-all.. but it seems not the right way. Does anyone have an idea how to solve without building own packages? Maybe this is an bug in kamailio Project? Many thanks for all proposals. BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 1

Re: [SR-Users] RTPProxy/Mediaproxy issue
by Daniel Grotti 25 Feb '14

25 Feb '14

Hi, Yes packet loss is very high. Please investigate around that. Daniel Ravi <wingsravi777(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 5

Problems while reading from location and parsing expires
by Jöran Vinzens 25 Feb '14

25 Feb '14

Hi All, we are upgrading from kamailio 3.1 to 4.1 and we got stuck at a Problem with usrloc module. While starting up kamailio reads location from DB and try to parse expires date from DB. this is the Problem: Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: <core> [db_ut.c:90]: db_str2int(): Unexpected characters: [-02-25 11:27:08] Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: <core> [db_val.c:75]: db_str2val(): error while converting integer value from string Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: db_mysql[km_row.c:68]: db_ mysql_convert_row(): failed to convert value Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: db_mysql [km_dbase.c:375]: db_mysql_fetch_result(): error while converting row #0 Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: usrloc [udomain.c:420]: preload_udomain(): fetching rows failed usrloc Module is configured with db_mode "1" so it saves it's entries by itself. When i have a look on the database it looks good and the expires field is an datetime. From logs it looks like it try to parse a unixtimestamp instead of the datetime. when i truncate the location table it starts perfectly, since there are no entries stored where it could fail parsing expires date. does anybody has similar issue or any suggestions how to solve? found in Version: 4.1.0 many thanks! BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 4

Re: [SR-Users] Kamailio as redirect
by Daniel Tryba 24 Feb '14

24 Feb '14

On Monday 24 February 2014 14:45:18 Mike Claudi Pedersen wrote: > okay i somewhat get that.. but cant i use something like a lookup in a > textfile instead of a db? You can. alias_db also accepts dbtext as backend: http://kamailio.org/docs/modules/stable/modules/db_text.html But TIMTOWTDI, alias_db was just a suggestion. Please keep the mailing list included. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

1 0

Re: [SR-Users] Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
by Fred Posner 24 Feb '14

24 Feb '14

On Mon Feb 24 10:50:07 CET 2014 Sean Kemball wrote: > New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, > and not a complete network idiot... but please be gentle. :) Welcome!s, > Questions: > > 1. Should the proposed topology, with Kamailio + an RTP proxy > behind a firewall, relaying to FS on an inside interface, work? > (Can't see why not) Yes, you said that your upstream is on the same private network. So it should be pretty straight forward. > 2. Does it need a local RTP proxy on the Kamailio box, particularly > if we turn off the ASA SIP inspect stuff? If you are all on the same private network, I would let FreeSWITCH handle the RTP, but you can do this a variety of ways. > 3. Can you recommend which RTP proxy to use? There seem to be at > least 3 that work with Kamailio. The box is CentOS 6.5, and it would > be nice to use known-to-work packages rather than compile from source. > (But eh, if I haveta). On your scenario, I'd just use FreeSWITCH for the media proxy. Again, many different ways to go here. > 4. Can anyone point me to some docs to explain what ports need to > be open between the Kamailio box and my upstream proxy/media server? > I can be more liberal between inside and DMZ I guess. Your upstream provider would generally tell you which rtp ports they would want opened. > 5. Is static NAT in this environment going to bite me, or should > it be OK? I've never had an upstream provider communicate with me on private nat. > 6. Is there any better documentation that we should be using to > make this easier, or should I just man up and try harder? Man up. =) Practive makes perfect. -- Fred Posner The Palner Group, Inc.

1 0

pcscf - removing registration entries from hash table
by Daniel Ciprus 24 Feb '14

24 Feb '14

Hello, Our current setup is Kamailio as IMS core and we're observing following behavior: Clients register themselves over TCP so pcscf creates hash table which keeps these information in memory. Once tcp reset occurs for any reason, registration entry stays in the memory and client re-registers himself with new pair ip:port. This kind of behavior creates problem because outbound SIP messages are being forked and requests are sent to non-existent tcp sessions which creates a lot of timeouts and confusion for AS. We're using ims_usrloc_pcscf module to keep reg entries in and as I noticed, with usrloc module, there was an option which automatically removed entry from hash table once tcp session was closed which is not the case in this module. Any suggestions how to get out of this ? thanks for any hints/suggestions. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

Kamailio as redirect
by Mike Claudi Pedersen 24 Feb '14

24 Feb '14

i want to setup a test environment to see how kamailio handles redirection. i want to be able to reroute calls between servers, where do i insert the configuration for this? can i append a list of user and destination to make the configuration go through, to see where the call is going? im new at kamailio, and im having a really hard time finding the right documentation to support a redirect server. could somone please give me an example? lets say i have 3 user 1000 1001 1002 and they are at 3 different servers eg. 1000(a)192.168.1.10 1001(a)192.168.1.11 1002(a)192.168.1.12 then i call user 1002 from user 1000, through the kamailio redirect server, which redirects to corresponding servers. Mike Claudi Pedersen

2 1

Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
by Sean Kemball 24 Feb '14

24 Feb '14

Hi New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, and not a complete network idiot... but please be gentle. :) We're trying to get Kamailio set up in front of a FreeSwitch-based SIP application server, to do some simple policy controls and, using one of the RTP proxy modules, to help handle the media side of things. Our SIP trunks come from our provider on a VLAN addressed as 10.x, and they provide an upstream proxy/media server (10.y address). The link also carries Internet traffic so we're running it through a Cisco ASA which breaks out the VLANs and static NATs the SIP stuff to our DMZ (10.z address). This is where we want to put the Kamailio box, where it should receive the calls and route them back through the ASA to the internal address (10.w range - all 10.w/x/y/z are separate, non-overlapping networks). Outbound calls from the application server (there are very few) should pass back up to Kamailio which will validate the numbers against an approved list (the app should only dial a subset of numbers) and pass them to the upstream proxy if necessary. We need to ensure that inbound calls from outside trombone through our application server so that the caller doesn't get billed for any calls, but this should be the default I think as that side of things is handled by FS (the app stands up a new call and bridges it to the incoming). This topology should allow us to use Kamailio to hide details of the FS app from the outside world, and make life easier for the app developer who should just send/receive to the Kamailio box with no further thought or complexity. If we put the FreeSwitch box in the DMZ where we want to put Kamailio, and then turn on SIP packet inspection in the ASA, calls flow but quality is poor for some callers. If we turn off SIP packet inspection, we get no audio - I can't find any clear Cisco documentation for this but I think the SIP inspection stuff in the ASA seems to handle RTP NAT fixups and the like too. But with no docs it may as well be magic, and I want to remove it if possible. With Kamailio in the DMZ and FS internal, we loosely followed http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc and hacked out the voicemail and Lua stuff, but couldn't get calls to flow, whether or not we used the SIP inspection on the ASA. We were confused why this example isn't define as WITH_NAT and doesn't use a local RTP proxy. Surely that would be needed? Questions: 1. Should the proposed topology, with Kamailio + an RTP proxy behind a firewall, relaying to FS on an inside interface, work? (Can't see why not) 2. Does it need a local RTP proxy on the Kamailio box, particularly if we turn off the ASA SIP inspect stuff? 3. Can you recommend which RTP proxy to use? There seem to be at least 3 that work with Kamailio. The box is CentOS 6.5, and it would be nice to use known-to-work packages rather than compile from source. (But eh, if I haveta). 4. Can anyone point me to some docs to explain what ports need to be open between the Kamailio box and my upstream proxy/media server? I can be more liberal between inside and DMZ I guess. 5. Is static NAT in this environment going to bite me, or should it be OK? 6. Is there any better documentation that we should be using to make this easier, or should I just man up and try harder? TIA Sean Volunteer for our Red Puppy street appeal to help puppies like Gordy become guide dogs. ############################################################################# This email, including any attachments, is intended solely for the addressee(s) It is confidential and may be legally privileged. If you are not the intended recipient, you must not copy, disclose, distribute or otherwise use it or the information in it. Please notify the sender at once and delete it from your system immediately. Any views or opinions expressed are solely those of the author and do not necessarily represent those of Blind Foundation. The Blind Foundation does not accept responsibility for any viruses or other malicious code that may be transmitted with this email. #############################################################################

1 0

Change from_uri
by arun Jayaprakash 24 Feb '14

24 Feb '14

Hello, I have a problem that my calls get rejected by he pstn gateway as my gateway is expecting a DID number instead of a 4 digit extension as the 'from user". Let me explain my situation: 1. I have set callfwd_busy in my user preference table. 2. In my failure route I check to see if callfwd_busy has been set, if it has then I execute the following script: if (t_check_status("486|408")) { $avp(dst_number) = $rU; if (avp_db_load("$avp(dst_number)", "$avp(callfwd_busy)")) { xlog("LOG: avp(callfwd_busy)=$avp(callfwd_busy), avp(dst_number)=$avp(dst_number)\n"); } xlog("LOG: Failure route with $rU $avp(dst_number) \n"); $rU = $avp(callfwd_busy); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; } When I call from extension 7004 to ext 7002 ( this has callfwd_busy set to a DID number) the trace looks as follows: U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 54.200.xx.yy:5060;branch=z9hG4bKee75.ab791c5.0;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.219.63;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. U 172.10.30.15:5060 -> 172.10.30.15:5080 INVITE sip:18455981234@64.136.xx.yy:5060 SIP/2.0. Record-Route: <sip:54.200.xx.xx;lr=on;ftag=1830267367;nat=yes>. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Contact: "7004" <sip:7004@71.252.xx.yy:1025>. Max-Forwards: 16. User-Agent: Grandstream HT701 1.0.4.8. Privacy: none. P-Preferred-Identity: "7004" <sip:7004@ajfmc1.myDomain.net>. Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 453. . v=0. o=7004 8000 8000 IN IP4 172.10.30.1554.200.xx.xx. s=SIP Call. c=IN IP4 172.10.30.1554.200.xx.xx. t=0 0. m=audio 6248262482 RTP/AVP 0 18 4 8 2 97 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16,32-36,54. a=nortpproxy:yes. a=nortpproxy:yes. # U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 503 Unavailable. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. It looks like my gateway is expecting to see a DID number in the from header. Can someone let me know how I can dot this? Thank you. Arun

3 4

Query about RTP flow
by Nandini madhu 23 Feb '14

23 Feb '14

Dear All, I have read about SIP and RTP packet flows but ended with a question about RTP packets flow logic between client-to client when the SIP session starts. I have a running Kamailio (V 4.0.4) server on ubuntu (12.04) platform which is also configured fro DNS and DHCP servers. And following are my test bed set-ups: Kamailio server (DNS+ DHCP+Kamailio) | Router <----------------------------> Switch<----------------------------> Router | | clientA <--------------------- RTP(audio/video) ? --------------------> client B Now when the SIP session established (SIP Packets have to pass through all the routers and switch to reach Kamailio server), RTP session will start between Client A and B. But What is meant by client to client here ? what is the RTP packets path between the clients to reach their destinations ? Is the RTP packets have to pass through Routers and switch to reach its destination client ? (i.e Client A--->Router 1---->switch -----> Router 2-----> client B) ? What if i connect both client A and client B to the same router ? PS: I have not configured any media-relays (RTPproxy/Media-proxy) Please help me in clarifying all this doubts. Anybody's help will greatly appreciate. Regards, Nandini

1 0

HTABLE update from DB
by Uri Shacked 23 Feb '14

23 Feb '14

It does not work...I tried with autoexpire=0, I tried with no autoexpires define, I Tried defining a number and see if the value is updated.... this is my table definition: modparam("htable", "htable", "A=>size=8,dbtable=aaaaaa;initial=0,dbmobe=0;") I remind you i need the memory to be updated from the DB every interval, and not the DB form the memory. thanks. >Declare the table without autoexpire. Check again the module README. >Regards, >Ovidiu Sas >On Feb 23, 2014 11:23 AM, "Uri Shacked" <ushacked at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >>* Hi, *>>>>* Following my issue with reloading data. I am thinking of a way to keep *>>* data updated in memory. *>>* I read the HTABLE module again and notice the "db_expires" option. *>>* It does not work for me... *>>>>* I try to set it so an item will expire in 60 sec, and when expired, it *>>* will be reloaded from DB and not deleted. *>>>>* Any ideas? *>>>>* Uri*

2 1

RTPProxy/Mediaproxy issue
by Wingsravi R 23 Feb '14

23 Feb '14

Dear Kamailio'ns, I have configured kamailio (V 4.0) with RTPproxy server. Audio/Video call establishments are just fine. With this i have some unknown behaviour in my set-up is that When i am running kamailio server with RTPproxy instance, i am experiencing Jitter, Latency, pixelled audio/video calls. But when i run only Kamailio server without RTPproxy service running, audio/video calls are just fine (no any jitter, latency kind of issues). I also tried with Mediaproxy (AG projects) server but the results are same as like in Kamailio/RTPproxy case. What could be the problem with this set-up ? How can i solve this issue ? Please anybody help me in resolving this problem. Any help will greatly appreciate. Regards. Ravi.

4 7

HTABLE update from DB
by Uri Shacked 23 Feb '14

23 Feb '14

Hi, Following my issue with reloading data. I am thinking of a way to keep data updated in memory. I read the HTABLE module again and notice the "db_expires" option. It does not work for me... I try to set it so an item will expire in 60 sec, and when expired, it will be reloaded from DB and not deleted. Any ideas? Uri

2 1

Using Kamailio as a websocket LB
by Luis Silva 22 Feb '14

22 Feb '14

Hi guys, Is it possible to use Kamailio as a Websocket LoadBalancer (transparent to the Websocket content, rather it's SIP or other protocol)? Many thanks, Luís

3 3

parallel forking and 30x
by Andreas Granig 21 Feb '14

21 Feb '14

Hi, What is the intended behaviour from a general point of view if you get a 1xx on one branch and a 30x on another branch for a parallel fork? >From my understanding, the 30x is a final reply, so it's kept in tm until all other branches are finished, then it enters failure-route with the highest error code (e.g. a 487 if you cancel the other branch). What I actually want is intercepting the 30x and send another INVITE to the Contact of the 30x in the failure route. This works fine if you only have one branch, but it doesn't enter failure route for 30x if you have another active branch (e.g. in ringing state). Would it be possible to create a new branch directly from the reply route, or force to enter failure route for a specific reply/branch? Andreas

3 5

Kamailio not starting
by Owais ul Haq 21 Feb '14

21 Feb '14

Hello, I have deployed Kamailio-3.1 on a fedora machine. And my database is placed on another windows 2008 server situated on the local network. Problem is when I run kamailio. I get the following error from logs. <core> [cfg.y:3416] : parse error in config gile /usr/local/kamailio-3.1/etc/kamailio//kamailio.cfg, line 311, column 31-35 : syntax error <core> [cfg.y:3416] : parse error in config gile /usr/local/kamailio-3.1/etc/kamailio//kamailio.cfg, line 311, column 31-35 : Invalid arguments Line 311 is kamailio.cfg is "modparam ("usrloc", "db_url", DBURL)" And 'DBURL' is defined as #! define DBURL "mysql://openser:5mil5My5ql@192.168.1.20/openser" Can some one please help me out in resolving this error. Secondly I am deploying my database on seperate server for the first time. I have deployed kamailio-3.1 already but with database placed on same machine. Can someone please guide what changings do I need to make in kamailio.cfg and kamctlrc file to make it work. I have made some as per my understanding. Maybe they are a cause of this error. Thanks, Regards, Owais

2 1

Information
by hfermi 21 Feb '14

21 Feb '14

Good morning I would like to receive more information about your platform web based voip. Many Thanks and Regards Henry Landi

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 21 Feb '14

21 Feb '14

Hi, We have found the root cause for the problem that was reported (refer below mail for details) in async module. Below is the brief description, - async_route("Resume", "1") - At time t, the async records are stored at slot 't+1' of async list. - Every second, async records from the async list are processed. - In function 'async_timer_exec', slot = ticks % ASYNC_RING_SIZE; since slot is based on the ticks and if the previous invocation of 'async_timer_exec' did not finish in time, the next tick is missed. Subsequent call to 'async_timer_exec' will be with t+2. Here slot t+1 will be missed and the same will be processed only during next cycle. That means the async records will be stored for the next 100 seconds before they are actually processed. TM module will drop those timed out transactions and hence the error "t_continue: transaction not found" Now we have modified the code in such a way that slot is incremented sequentially irrespective of the ticks that is sent to the function 'async_timer_exec' function. This way we do not see any call failures and all our load runs are successful. If someone is interested we can share the code as well. Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Thursday, January 23, 2014 12:18 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, >From our repeated load tests what we can conclude irrespective of the number of simultaneous calls, there is always this error "t_continue: transaction not found" occurring. If I run say 20 cps, then after running 5000 calls, we observe exactly 20 calls failing with the above error. We doubt that there is something happening during a particular point in time (for a second) which impacts the saving of those new transactions into shared memory. For 10 cps run, we observe exactly 10 call failures. We repeated with different cps and found that error is exactly equal to the cps being run. Any configuration we are missing. Anyone can help? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

Call Control for Credit and Rate
by Raj Roy Ghandhi 21 Feb '14

21 Feb '14

Hi, I want to implement the call control by 1. Validating the CLI and get the credit from DB 2. Check the Rate for the dialed number 3. set the call timeout for the call duration. ( credit/rate) 4. pass the call to remote PSTN GW Please guide me on this. Regards, Roy.

3 2

Debian packages not updated since august 2013?
by Johan Wilfer 21 Feb '14

21 Feb '14

Hi, Looking at http://www.kamailio.org/wiki/packages/debs the suggestion is to use http://deb.kamailio.org/ for debian packages. But the last version of kamailio 4.0 here is 4.0.3, and 4.1 doesn't exist. The most files are last modified in 2013-08.. Here for example: http://deb.kamailio.org/kamailio/dists/wheezy/main/binary-amd64/ Is there a new server to get the deb-packages from or should I build from source from now on? /Johan

3 3

Performance impact with AVP and VAR's
by Jijo 21 Feb '14

21 Feb '14

Hi All, I have around 300 AVP's and quite amount of VAR's are used in the config file? Does that impact performance?, If so how can i improve it? Thanks Jijo

4 5

Re: [SR-Users] Issue with Async module
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, thanks for reporting, I pushed a slightly different patch in master, 4.1 and 4.0 branches -- free is done even when ai->act is null. Cheers, Daniel On 20/02/14 15:13, Surendra wrote: > Hi Daniel, > > > I am using async module to continue the transaction in > asynchronously. But I am observing a leak in shared memory. Then i checked > the code, we are not freeing the async_item_t after calling t_continue. > Same this has checked with MEMDBG=1. > > Original code: > while(1) { > lock_get(&_async_list_head->ring[slot].lock); > ai = _async_list_head->ring[slot].lstart; > if(ai!=NULL) > _async_list_head->ring[slot].lstart = ai->next; > lock_release(&_async_list_head->ring[slot].lock); > > if(ai==NULL) > break; > if(ai->act!=NULL) > tmb.t_continue(ai->tindex, ai->tlabel, ai->act); > } > > > Changed code: > > while(1) { > lock_get(&_async_list_head->ring[slot].lock); > ai = _async_list_head->ring[slot].lstart; > if(ai!=NULL) > _async_list_head->ring[slot].lstart = ai->next; > lock_release(&_async_list_head->ring[slot].lock); > > if(ai==NULL) > break; > if(ai->act!=NULL) > { > tmb.t_continue(ai->tindex, ai->tlabel, ai->act); > shm_free(ai); /* Added by surendra */ > } > } > > Please correct this if I am wrong. > > > Regards > @surendra > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs
by Neo Quartz 20 Feb '14

20 Feb '14

Hi All, I installed Kamailio server on my ubuntu machine. And installed rtpproxy also in the same machine. I didm't make many changes to default configuration files, kamctlrc and kamailio.cfg. Now I am able to communicate between two SIP clients, if both sip clients are in the same LAN. But I am not able to communicate between one sip client in one LAN and the other SIP client in other LAN. Sip Call handling and ringing etc. is ok but there is no voice for this case. I don't have much experience in routing for NAT traversal. I tried to experiment with rtpproxy_manage function in kamailio.cfg file. But I was not successful. Please help me by providing any hints or pointers to proceed further. I am using only Kamailio and Rtpproxy. No other software like Asterisk or FreeSwitch. ************* These are defines I used in kamailio.cfg file: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PRESENCE listen ip address changed ******************* Kamailio is compiled with following modules: include_modules= db_mysql dialplan presence presence_xml ******** Regards, Sateesh

2 1

understanding the sqlops module and kamailio children
by Uri Shacked 20 Feb '14

20 Feb '14

Hi, Following my issue with the above.... 1. Is there a way to see if a worker (child) is processing a transaction for too long (let's say 2-3 sec)? 2. Is there an option to see if the DB connection is alive? (I am familiar with db_mysql options) 3. If all workers are stuck on this "long" transaction, is there a Q for the new INVITES? Can I see how many are there? What I was experiencing is that when my DB was very slow, kamailio crashed after a while.... I will configure db_mysql to ping every 30 sec and timeout after 1 sec. Still, it can create a 3 sec delay for transactions. Any one has an idea? Using the ASYNC module can help here? Does it free the worker to deal with new requests? BR, Uri

2 1

Asterisk Ignore "UNREGISTER" because retransmit
by pavel＠eremina.net 20 Feb '14

20 Feb '14

Hi, All! Try to register user in Kamailio and then relay REG massage to Asterisk. Asterisk register user and looks good. But when I send Unreg message to Kamailio, it unregister user successfully but Asterisk say: Message REGISTER Ignoring because retransmit (CSeq 1234 our CSeq 1234). Did someone know how solve this problem: i think that Asterisk need different CSeq for first REGISTER and Second after 401 code. dump my conf: At route[REGISTER]: if (!save("location")) { sl_reply_error(); } else { reg_fetch_contacts("location", "$tu", "cc"); $var(useragent) = $ulc(cc=>user_agent); xlog("L_INFO","EXpires in Contact: $ulc(cc=>expires)"); if ($ulc(cc=>expires)==0) { $var(query) = "insert into activity values (now(),'unr','"+$ci+"','"+$fu+"');"; sql_query("events", "$var(query)"); $var(string_temp) = "$(hdr(expires))"; xlog("L_INFO", "UNregistered $ru FROM $fu exp $var(string_temp)"); } else { if ($var(registered)==0) { $var(query) = "insert into activity values (now(),'reg','"+$ci+"','"+$fu+"');"; sql_query("events", "$var(query)"); } xlog("L_INFO", "Registered $ru FROM $fu exp $ulc(cc=>expires)"); } } xlog("L_INFO","DU(TD): $td"); if ($td == "mydomain.com") {route(TOASTERISK); } exit; } } route[TOASTERISK] { xlog("L_INFO","ROUTE to ASTERISK from $ru to $tu"); #append_branch(); t_on_failure("FAILAUTH"); $ru = "sip:"+ $td; consume_credentials(); route(RELAY); exit; } failure_route[FAILAUTH] { if (t_is_canceled()) { exit; } if (t_check_status("(401)|(407)")) { xlog("L_INFO", "Remote asked for authentication"); $avp(realm)=""; uac_auth(); } route(RELAY); }

1 0

Issue with Async module
by Surendra 20 Feb '14

20 Feb '14

Hi Daniel, I am using async module to continue the transaction in asynchronously. But I am observing a leak in shared memory. Then i checked the code, we are not freeing the async_item_t after calling t_continue. Same this has checked with MEMDBG=1. Original code: while(1) { lock_get(&_async_list_head->ring[slot].lock); ai = _async_list_head->ring[slot].lstart; if(ai!=NULL) _async_list_head->ring[slot].lstart = ai->next; lock_release(&_async_list_head->ring[slot].lock); if(ai==NULL) break; if(ai->act!=NULL) tmb.t_continue(ai->tindex, ai->tlabel, ai->act); } Changed code: while(1) { lock_get(&_async_list_head->ring[slot].lock); ai = _async_list_head->ring[slot].lstart; if(ai!=NULL) _async_list_head->ring[slot].lstart = ai->next; lock_release(&_async_list_head->ring[slot].lock); if(ai==NULL) break; if(ai->act!=NULL) { tmb.t_continue(ai->tindex, ai->tlabel, ai->act); shm_free(ai); /* Added by surendra */ } } Please correct this if I am wrong. Regards @surendra

1 0

Problem with special sip notify (trying to make central provisioning for endpoints)
by Pedro Niño 20 Feb '14

20 Feb '14

Hi, how are you all I have a lab setup for using Kamailio and asterisk, with realtime, and currently I am facing 2 problems. The first one, is when trying to get the phones to reboot. Asterisk send a notify with the message, reboot. But the phone answers with a 401, and asks for a digest. In a normal asterisk environment it would not be a problem, asterisk tries to answer with the user and password cipher with the digest. But because of using this design, asterisk sends the notify with the user and blank password (I am using realtime, so peers can register from kamailio with no password).of course this makes that the phone refuses, as the notify doesn't have the right password. Is there any way to make kamailio intercept the notify sent from the phone and answer with the right credentials? _------------------- Second: When having the users registered in the same scenario, and get an asterisk 'reload', and the users try to make a call before they re-register, only the first call make it thru Of course, as kamailio is in the middle, the users doesnt notice until they try to make a call. In asterisk, I keep seeing the kamailio trying to register and getting a 404. This keeps happening until I sent a sip reload at the asterisk side and wait for the users to register before making calls. Thanks for your time....

1 0

8 digit restriction on dialplan
by phillman25 20 Feb '14

20 Feb '14

Dear List I am trying to restrict 8 digit dial on my dialplan after the prefix 6859A , could someone help me with the correct syntax? Please see below what i have: Dpid: 1 Pr: 1 Match Op: 1 Match Exp: sip:6859A(only 8 additional digits allowed).* Match Len: 0 Subst Exp: sip:6859A(only 8 additional digits allowed).* Repl Exp: 3005A1883(a)xx.xx.xx.xx So basically any 8 digits after the prefix 6859A is allowed otherwise there shouldn't be a match, how could i go about doing this? Thanks in advance Phillip

1 0

Error in carrierroute in kamailio 4.1
by Eduardo Alonso 20 Feb '14

20 Feb '14

Hi all: It's my first mail to the community and I know that will be a great community to share knowledge. OS Debian 6.4 Kamailio 4.1 I have the same problem that Oliver Roth describes in http://lists.sip-router.org/pipermail/sr-users/2013-October/080096.html, but in Kamailio 4.1: Module carrierroute not working starting from version kamailio 4.1 trying to activate carrierroute in kamailio.cfg loadmodule "carrierroute.so" i get the following error when starting kamailio (5794) ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/carrierroute.so>: /usr/local/lib64/kamailio/modules/carrierroute.so: undefined symbol: cfg_set_error_function 0(5794) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-28: failed to load module ERROR: bad config file (1 errors) Any idea? Maybe him o someone has the solution. Thanks all for your great work. Best regards. -- *Eduardo Alonso Gil* VoIP Systems Engineer @ Quobis <http://www.quobis.com/> | e: eduardo.alonso(a)quobis.com | t: +34902999465 Visit our booth at *MWC Barcelona*. CS60 Spanish Pavilion. More info<http://www.quobis.com/index.php?option=com_content&task=view&id=236&Itemid=…>

1 0

Issues with keepalive_timeout
by Kristian Kielhofner 20 Feb '14

20 Feb '14

Hello All, I'm trying to use keepalive_timeout with 4.1.1 as described here: http://lists.sip-router.org/pipermail/sr-users/2012-August/074337.html I believe I've followed these configuration steps and those defined in the nathelper module. Here are the relevant portions of my config: modparam("nathelper", "natping_interval", 60) modparam("nathelper", "keepalive_timeout", 180) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_from", "sip:pinger@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 9) And of course a setting the correct branch flag (verified) during REGISTER handling: setbflag(9); My goal is to use SIP OPTIONS messages to verify the reachability of all of my endpoints (not just NATd ones). I would just lower my re-registration timeout but there are other issues associated with that kind of change. With my current configuration (after a successful registration) a SIP OPTIONS message is sent and replied to: U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-9627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-93072f43-7c4cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.2:5060 -> 4.2.2.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 4.2.2.1:5060;received=4.2.2.1;branch=0. Call-ID: 8af79586-93072f43-7c4cf55(a)4.2.2.1. From: <sip:pinger@domain.com>;tag=uloc-5303810e-7c78-1-2273f0eb-9627292. To: <sip:user372@4.2.2.2:5060>;tag=0;ob. CSeq: 1 OPTIONS. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain. Supported: replaces, 100rel, timer, norefersub. Allow-Events: presence, message-summary, refer. User-Agent: My_UA. Content-Length: 0. . If I kill the registered UA (making it unavailable) Kamailio endlessly sends OPTIONS messages, never receives a reply, and never removes the registration from usrloc: U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-b627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-b3072f43-f35cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-c627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-c3072f43-b75cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-d627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-d3072f43-7b5cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-e627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-e3072f43-3f5cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . (and on, and on, and...) Eventually the registration timeout kicks in, the usrloc entry is removed, and Kamailio stops sending OPTIONS to the (long) unreachable endpoint. Is this expected behavior? What am I doing wrong? Thanks! -- Kristian Kielhofner

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Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, setting the hash size to 32MB is way too much, are you having like 100 000 000 active calls at the same time? The size of the hash is used for computing the number of slots in the hash table, not the capacity in terms of memory size of all items stored in hash table. Cheers, Daniel On 03/02/14 07:33, Shankar wrote: > > Hi Alex, > > We set the hash size to 32 MB and found that the hash size is not the > same as configured one. At the max it was taking 64KB. > > Regards, > > Shankar > > Date: Mon, 03 Feb 2014 00:27:29 -0500 > > From: Alex Balashov <abalashov(a)evaristesys.com > <mailto:abalashov@evaristesys.com>> > > To: sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > Subject: Re: [SR-Users] Regd. MAX_LDG_LOCKS > > Message-ID: <52EF28C1.5010204(a)evaristesys.com > <mailto:52EF28C1.5010204@evaristesys.com>> > > Content-Type: text/plain; charset=windows-1252; format=flowed > > On 02/03/2014 12:22 AM, Shankar wrote: > > > Also we found that in ?dialog.c? the variable dlg_hash_size should be > > > ?unsigned int?. Hope this will be taken care in next release of > Kamailio. > > How did you arrive at that conclusion? > > -- > > Alex Balashov - Principal > > Evariste Systems LLC > > 235 E Ponce de Leon Ave > > Suite 106 > > Decatur, GA 30030 > > United States > > Tel: +1-678-954-0670 > > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, February 03, 2014 10:53 AM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* Regd. MAX_LDG_LOCKS > > Hello, > > Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? > > Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if > the hash size is increased? > > Also we found that in ‘dialog.c’ the variable dlg_hash_size should be > ‘unsigned int’. Hope this will be taken care in next release of Kamailio. > > Regards, > > Shankar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, it is not a really necessary to increase the value of the locks. For better performances you can do it, but again, it is not a must. Cheers, Daniel On 04/02/14 06:50, Shankar wrote: > > Hi, > > Can someone clarify here please? > > Regards, > > Shankar > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, February 03, 2014 10:53 AM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* Regd. MAX_LDG_LOCKS > > Hello, > > Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? > > Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if > the hash size is increased? > > Regards, > > Shankar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio 4.1 Installation from Source
by Francesco Maria Magnini 19 Feb '14

19 Feb '14

Hi, following the documentation, the following lines need to be updated on the init.d script. DAEMON=/usr/local/sbin/kamailio CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Anyway, there is no CFGFILE already setted. Is it correct to add CFGFILE along with DAEMON? On top of this, in the check_fork () procedure, there’s a reference to a cfg_file that is nonexistent. In my installation i changed it to the real path of the kamailio.cfg file generated during installation. check_fork () { if grep -q "^[[:space:]]*fork[[:space:]]*=[[:space:]]*no.*" /usr/local/etc/kamailio/kamailio.cfg; then log_failure_msg "Not starting $DESC: fork=no specified in config file; run /etc/init.d/kamailio debug instead" exit 0 fi } Is it correct to change path in the check_fork procedure with the correct one? Regards,

2 2

Kamailio World - Presentations
by Daniel-Constantin Mierla 19 Feb '14

19 Feb '14

Only very few presentations accepted at Kamailio World Conference & Exhibition are not yet published. In a matters of days we should have the draft of the schedule published. It is again a good variety of topics around Kamailio and other open source VoIP applications, presenting experiences from the field, showing new product and solutions or open debate and demos panels. You can lean about various ways of deploying scalable platforms with Open Stack, Erlang or asynchronous, carrier grade charging systems, managing security in VoIP deployments or how to do real time troubleshooting. More details about presentations are available at: - http://conference.kamailio.com/k02/schedule/ Information about speakers can be found at: - http://conference.kamailio.com/k02/speakers/ In order to allow a fair enough time to see what the content of the event is going to be, the early registration deadline has now been set to this Friday, February 21. Don't miss the chance to learn from about 30 sessions and meet more than one hundred professionals in real time communications and open source technologies: - http://conference.kamailio.com/k02/registration/ Looking forward to meeting you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

uac_reg_request_to
by Isaac McDonald 19 Feb '14

19 Feb '14

Hello All, I need some clarification on the function uac_reg_request_to. The doc says "This function can be used to send an authenticated request to a remote user in the uac registrations table" Does this mean I can forward a SUBSCRIBE request, or any request for that matter, to an upstream server that's Kamailio is registered to via the UAC module? Secondly, I'm having trouble matching the user on this function. The error is: "ac [uac_reg.c:996]: uac_reg_request_to(): no user: 2503" User 2503 does exist in the uac_reg table in column "l_username". What if I want to match by UUID instead? How does that get mapped back to a local user? Thanks -- Isaac McDonald SusCall, LLC E-Mail: imcdona(a)suscall.com

1 0

Memory leak in http_query() function (utils module)
by Guillaume Bour 18 Feb '14

18 Feb '14

Hi folks, We identified a memory leak in *utils* module, when calling the http_query() function. a memory piece is allocated with malloc() to store http request result string (functions.c:54), but this memory is never freed. There is also a possible crash issue, because curl_easy_getinfo() is called *after* curl_easy_cleanup(), which must not be done (see http://curl.haxx.se/libcurl/c/curl_easy_cleanup.html) I attached patches to fix those issues (applies to 4.0 branch) Regards, Guillaume Bour

2 5

error using siremis
by Rubi Mazaki 18 Feb '14

18 Feb '14

Hi I successfully install siremis 4.0.0 +kamailio 4.1 on my ubuntu (32 bit) server. Kamailio was installed successfully and I can now run "kamctl moni" and see a nice status screen. I had to setup again siremis (by deleting the install.lock and opening the localhost/siremis setup again) but one that was achieved I can now login in to the siremis site without errors. Now to the problem: Once I enter "groups" and press "add" I get the following error displayed: [2014-02-12 20:46:33 (GMT)] An exception occurred while executing this script: Error message: #8192, preg_replace(): The /e modifier is deprecated, use preg_replace_callback instead Script name and line number of error: /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compiler.class.php:270 *function:* errorHandler ( 8192, "preg_replace(): The /e modifier is deprecated, use preg_replace_...", "/home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compi...", 270, Array(8) ) @ /home/rubi/siremis-4.0.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 8192, "preg_replace(): The /e modifier is deprecated, use preg_replace_...", "/home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compi...", 270, Array(8) ) @ *function:* preg_replace ( "~\{\*(.*?)\*\}|\{\s*literal\s*\}(.*?)\{\s*/literal\s*\}|\{\s*php...", "'{php' . str_repeat(" ", substr_count('\0', " ")) .'}'", "<form id="{$ form.name}" name="{$form.name}"> <div style="paddin..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compiler.class.php 270 *function:* _compile_file ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "<form id="{$form.name}" name="{$form.name}"> <div style="paddin...", Null ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1484 *function:* _compile_source ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "<form id="{$form.name}" name="{$form.name}"> <div style="paddin...", Null, "/home/rubi/siremis-4.0.0/siremis/themes/default/template/cpl/%%2..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1417 *function:* _compile_resource ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "/home/rubi/siremis-4.0.0/siremis/themes/default/template/cpl/%%2..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1256 *function:* fetch ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail...." ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/FormRenderer.php 95 *function:* renderSmarty ( Object(EasyForm), "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail...." ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/FormRenderer.php 45 *function:* render ( Object(EasyForm) ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/EasyForm.php 1648 *function:* renderHTML ( ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/EasyForm.php 1538 Can anyone tell me what I did wrong, or how can I re-install completely siremis? Hopefully that's not a bug :/ With Regards, Rubi Mazaki Email: rubimazaki(a)gmail.com

2 1

One more question 'bout db_mode
by Victor V. Kustov 18 Feb '14

18 Feb '14

Hi all! my usrloc database use postgress. In config I had: modparam("usrloc", "db_mode", 1) after day of work I see strange thing: # kamctl ul show --brief | grep -c AOR 26 psql -U kamailio -c "select * from location;" ... (1 row) 26 registered users in inner kamailio's memory and only 1 in database. See in doc: 1 - Write-Through scheme. All changes to usrloc are immediately reflected in database too. This is very slow, but very reliable. Use this scheme if speed is not your priority but need to make sure that no registered contacts will be lost during crash or reboot. I think saved locations in inner memory database must be identical. Am I wrong? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.12.9-calculate GNU/Linux

1 0

Change table type...
by arun Jayaprakash 18 Feb '14

18 Feb '14

Hello, I am planning to change the table type in the Kamailio Mysql db from MyISAM to Innodb. Could anyone let me know if this will be an issue? Thank you. Arun

2 2

modify To header, prefix + ip, using dispacher module.
by Helena Garcia-Nieto 17 Feb '14

17 Feb '14

Hey, Thank you in advanced for the time taken on my question! I have an environment phone---sbc---kamailio---gw1, gw2 I am using kamailio 4.2.0, quite basic config, using dispatcher module to route to the gw and to monitor them with options. I have a file dispatcher.list with the two gw ips: # $Id$ # dispatcher destination sets # # line format # setit(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # gateways 1 sip:10.X.X.38:5060 IP 1 sip:10.X.X.39:5060 AP I have to modify the TO header, it must be prefixed and gw ip must be placed there (gw requirements) I use on the routing function ds_select_domain("1", "8") t_on_failure("RTF_DISPATCH"); and just before the ROUTE(RELAY) I have: # Add profix to gw: 0199 subst_uri('/^sip:(.*)/sip:0199\1/i'); # add prefix to RURI subst_hf("To", "/(.*):(.*)@(.*)$/\1:0199\2@$rd>/", "a"); #add prefix to TO, modify ip on TO use ruri ip It works fine if the first gw is available, but if the first option (which is the last gw on the list) is not accessible, it waits for the 30s and then tries the second one, but the ip on the TO header is not correct anymore, since it is the first gw and not the second. Question: why does it wait until the timeout if it is monitoring it with option and seeing it is not reachable? I tried modifying again the TO header with: subst_hf("To", "/(.*)@(.*)$/\1@$rd>/", "a") before the route(relay) on the failure routine, but it does not work properly, I am getting strange TO header like: To: <sip:0199XXXX@10.X.X.39> <sip:XXXXXX@10.X.X.38> It looks like it is not parsing the to header it has before but adding a parsed version of the original header at the end of the other one.... I tried with subst as well but the result is more or less the same. Probably there is a better way to set the prefix and the correct IP on the to header... I'll appreciate any help with this issue. Thanks again for your time Helena Helena Garcia-Nieto de Mir Morodo Group VoIP Department E-mail: helena.gnieto(a)morodo.co.uk Morodo Limited voip(a)morodo.co.uk / www.mo-call.com Place of Registration: Companies House, England & Wales / Registered Office: 7 Breasy Place, 9 Burroughs Gardens Hendon, London NW4 4AU

4 12

deb.kamailio.org down?
by Paul Belanger 17 Feb '14

17 Feb '14

Anybody know why deb.kamailio.org is down? Is there a mirror setup some place else? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 2

Re: [SR-Users] Msilo message body type
by sipatse＠in.gr 16 Feb '14

16 Feb '14

2 1

Kamailio call forward help
by Sajan Thomas 16 Feb '14

16 Feb '14

Hi sir, I have 2 server running with kamailio.I have tested in both with creating two users and chat using jitsi client. Please help me with 1.How can i implement kamailio call forward? suppose if make achat or call to kamailio server on user shoud can access the same from kamailio server 2. 2.Is there any possible modules are available for kamailio Please help me i am really got stuck here. Expecting your kind reply.

1 1

Error with ACK from SIPML5
by Jayesh Nambiar 14 Feb '14

14 Feb '14

Hello All, I have a setup which is SIPML5 -> Kamailio(Websocket) -> Freeswitch. All I am trying to do is initiate a call from SIPML5 and Play a Music file from freeswitch. This works, but the call hangs up after 30 seconds due to ACK timeout from from the SIPML5. What I can see is the SIPML5 from Chrome does send an ACK on websocket, but I see the following error in kamailio syslog: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [msg_translator.c:1725]: build_req_buf_from_sip_req(): could not create Via header* *ERROR: <core> [forward.c:607]: forward_request(): ERROR: forward_request: building failed* *ERROR: sl [sl_funcs.c:371]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)* Can someone help me understand what I am missing here and why kamailio is not able to create a VIA header to forward this request towards freeswitch. I have built a simple config file by following https://gist.github.com/jesusprubio/4066845 as reference. Any help in understanding this is really appreciated. Thanks, --- Jayesh

4 11

websocket.so module not generated
by Alvaro Urionabarrenetxea Gomez 14 Feb '14

14 Feb '14

Hi. I have a problem when I'm trying to start with my Kamailio server. I have an error about not existing "websocket.so" module. Here is the error: root@kamailio:# /etc/init.d/kamailio start ** Not starting Kamailio SIP server: invalid configuration file!* *** ** 0 (8327) ERROR: <core> [sr_module.c:587]: load_module(): ERROR: load_module: could not find module <websocket> in </usr/local/lib/kamailio/modules/>* *0(8327) : <core> [cfg.y:3408]: yyerror_at(): parse_error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-25: failed to load module* *ERROR: bad config file (1 errors)* *** And here is the line 318 in kamailio.cfg: *loadmodule "websocket.so"* I looked for that module in the correspondent folder but isn't there. Can anyone upload the websocket.so file to another mirror, MEGA for example? I tried to generate another time, download another kamailio versions but the error stills there. Much apreciated. Thanks!

1 0

FW: Client site NAT negotiation with no STUN.
by Rob Moore 14 Feb '14

14 Feb '14

Hi there. So I'm fighting what must be a bit of a common problem here but the usual googling isn't really helping me out. A bit of background: Classic Nat scenario of a sip handset behind a NAT firewall over the internet to VoIP SBC setup using Kamailio 3.3 and RTPProxy and Freeswitch as the registrar / calls server behind kamailio. (please excuse the beautiful ASCI art) -------------- ---- ----| RTPproxy |------- --------------- ------------ | -------------- | --------------- |SIP Phone|--------|Firewall |---------internet-------------| kamailio |-----------|Freeswitch| --------------- ------------ -------------- --------------- SIP Phones can register fine (over TCP or TLS, Kam offloads TLS if used) however when establishing a call the SIPPhone sends its local network IP address within the SDP instead of the firewalls external address. Causing media to me miss directed by the RTP Proxy. I've put a phone in with STUN to check that this is the only issue and that phone works fine .Usually in this situation we would just use STUN or TURN / ICE to get around the issue however the handsets that we are working with (and have to work with for other reasons) do not support any of these technologies. So I've been forced to try and find another solution. I was hoping to find some method within kamailio / RTPproxy to resolve this. I could just rewrite the SDP in certain situations but that feels a little bit brutal if I'm honest. I've heard of RTPproxy being able to do some sort of rtp latching where it can correct its destination ip address after it receives its first rtp packets from the SIP Phone after being forwarded by the onsite firewall but other than this 'word of mouth' suggestion I've not managed to find any detail of this functionality in the usual online resources. Can anyone suggest any other potential methods for getting around this problem? Many Thanks Rob

2 3

SSL connection issue
by Owais ul Haq 14 Feb '14

14 Feb '14

Hello, I have installed kamailio server. Now I am trying to use SSL for my server - client connections. I have made certificates succesfully. Now when my client connects, I get the following error. [tcp_read.c:882] : ERROR : TLS accept : error : 1408F10B : SSL routines:SSL3_GET_RECORD:wrong version number I have tried finding solution online. I am asked to change the version number. Can some one please tell me where I have to change the version number ?? And change it to what ?? Thanks. Cheers, Owais

2 2

Problem with SSL version number
by Owais ul Haq 14 Feb '14

14 Feb '14

Hi kamailians, I am trying to connect kamailio and use TLS between kamailio and client. I could not get any tls_method to work. Openser error messages for each method are listed below: with tls_method=SSLv2: " tls_accept: Error in SSL: 3(8332) tls_error: error:140EC0AF:SSL routines:SSL2_READ_INTERNAL:non sslv2 initial packet " with tls_method=SSLv3 or tls_method=TLSv1: " 3(8386) tls_accept: Error in SSL: 3(8386) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number " With tls_method=SSLv23: " 3(8403) tls_accept: Error in SSL: 3(8403) tls_error: error:140760FC:SSL routines:SSL23_GET_CLIENT_HELLO:unknown protocol " I am making these changes in tls.cfg. Can someone please tell what changes are required ?? Thanks Cheers, Owais.

2 1

Re: [SR-Users] Suggestion for VOIP Solution
by DanB 14 Feb '14

14 Feb '14

That is not entirely true. We use MySQL without any kind of issues via ODBC driver. DanB On 14.02.2014 10:18, sr-users-request(a)lists.sip-router.org wrote: > Asterisk would be suitable solution, FREESwitch is also fine, but with > FREESwitch latest version uses postgreSQL, and removed support for MySQL, > if you are comfortable with postgreSQL for reporting stuffs like that then > you can proceed with FREESwitch. > > Regards > > > > Thank you with regards, > Gopalkrishnan N. > Mob: +91 99404 91346 > VoIP call -sip:saigop@gtalk2voip.com

1 0

Suggestion for VOIP Solution
by Vijay Thakur 14 Feb '14

14 Feb '14

Hi All List Members, I have worked on kamailio for one year. It is a great open source Skype like solution. Now i have to deploy a new solution with new requirements. Kindly suggest that what will be the best open source application will fulfill my requirements. I need the following facilities: (1) Call Recording (Outbound/Inbound) (2) On Demand Recording (3) Dictation Service (4) Complex Inbound Routing (5) IVR (Interactive Voice Response) (6) Auto Attendant (7) Client Recordings Mgmt & Playback Portal (8) Multi-level Client hierarchy / Subscription Plans (9) Voicemail (10) Time based Routing (Message Box) (11) Call Queuing & Conferencing (12) Bulk Download & Emailing (13) Call Whisper (14) Smartphone apps (SIP clients) Regards, /

5 4

understanding the sqlops module and kamailio children.
by Uri Shacked 13 Feb '14

13 Feb '14

Hi, Lately i am experiencing some issues with the sqlops module. let's say, for example, that i use sqlops to query some information about the subscriber when an INVITE arrives. I use kamailio 3.3.2, MySQL 5.6 and the DB server is remote and not local. Usually, the DB result is very fast and the call continues. But, when the DB is unavailable, or slow, the result is delayed or lost. In this case: 1. What is the query time out in kamailio? 2. If the database is available but slow, does it mean that no new INVITE will be dealt by the current child process until the sqlops is gone? 3. does anyone has a good idea for managing sqlops in a call? Thanks, Uri

2 2

How do I allow any user to register? And why doesn't a call reach the other end in my case?
by Victor Sergienko 13 Feb '14

13 Feb '14

Hi, I'm pretty ignorant about Kamailio, sorry for it and possible terminology mistakes. I'd like to allow any user to register. Ideally, I'd like to make some script (Lua/whatever) function to check some data encoded in a username and allow registering based on that username only. I undefined WITH_AUTH in kamailio.cfg, and it used to work for me; but sometimes I'm starting to get "SIP Progress: 100 trying -- your call is important to us ()" and the call never reaches the other party. Here's a sample of my logs during the failed call: https://gist.github.com/singalen/c343f09b67dcf7731fe0 What is the problem here? I start and stop the clients all the time, so there might be some garbage registrations accumulated. I assume I also need a user registration to discard all his previous registrations? Thank you!

1 0

$var binary operation
by Jijo 13 Feb '14

13 Feb '14

Hi I tried the binary operation using $var as per the documentation and its seems not working. Can somebody help me to understand it. http://kamailio.org/dokuwiki/doku.php/pseudovariables:3.1.x $var(a) = 3 + (7&(~2)); if( [ $var(a) & 4 ] ) { xlog("var a has third bit set\n"); } The format with square bracket in the conditional statement is showing parse error when i load the cfg. So it tried like below and that too not working $var(a) = 8; if( $var(a) & 4 ) { xlog("var a has third bit set\n"); } Please let me know if something missing here.? Thanks Jijo

2 2

core generated - 502 Bad gateway
by Tuan Viet Nguyen 13 Feb '14

13 Feb '14

Hello, Upon receipt an 502 Bad gateway, I have the following error then Kamailio crashed Feb 11 11:49:57 kamailio23 /usr/local/sbin/kamailio[28895]: WARNING: tm [t_lookup.c:1564]: t_unref(): WARNING: script writer didn't release transaction Feb 11 11:50:03 kamailio23 /usr/local/sbin/kamailio[28926]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 31 Feb 11 11:50:03 kamailio23 /usr/local/sbin/kamailio[28876]: ALERT: <core> [main.c:788]: handle_sigs(): child process 28903 exited by a signal 11 Feb 11 11:50:03 kamailio23 /usr/local/sbin/kamailio[28876]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated (gdb) bt #0 0x000000000053a81d in qm_free (qm=0x7fd6f962a000, p=0x75d63ff100000000) at mem/q_malloc.c:476 #1 0x00007fd700c5581c in free_cell (dead_cell=0x7fd6f9777bf0) at h_table.c:162 #2 0x00007fd700c56783 in free_hash_table () at h_table.c:443 #3 0x00007fd700c6981b in tm_shutdown () at t_funcs.c:122 #4 0x00000000004ee6f0 in destroy_modules () at sr_module.c:790 #5 0x0000000000463baa in cleanup (show_status=1) at main.c:573 #6 0x0000000000464c32 in shutdown_children (sig=15, show_status=1) at main.c:715 #7 0x0000000000466223 in handle_sigs () at main.c:806 #8 0x000000000046a459 in main_loop () at main.c:1767 #9 0x000000000046ca85 in main (argc=5, argv=0x7fff2f13ec38) at main.c:2566 (gdb) bt full #0 0x000000000053a81d in qm_free (qm=0x7fd6f962a000, p=0x75d63ff100000000) at mem/q_malloc.c:476 f = 0x75d63ff0fffffff0 size = 4880 next = 0x7fd6f98c5380 prev = 0x7fd6f9634840 __FUNCTION__ = "qm_free" #1 0x00007fd700c5581c in free_cell (dead_cell=0x7fd6f9777bf0) at h_table.c:162 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x59e92da4 cbs = 0x0 cbs_tmp = 0x7fd6f98c5370 __FUNCTION__ = "free_cell" #2 0x00007fd700c56783 in free_hash_table () at h_table.c:443 p_cell = 0x7fd6f9777bf0 tmp_cell = 0x7fd6f98fb9c0 i = 3084 #3 0x00007fd700c6981b in tm_shutdown () at t_funcs.c:122 __FUNCTION__ = "tm_shutdown" #4 0x00000000004ee6f0 in destroy_modules () at sr_module.c:790 t = 0x7fd7022322f0 foo = 0x7fd702231cc0 #5 0x0000000000463baa in cleanup (show_status=1) at main.c:573 memlog = 32727 __FUNCTION__ = "cleanup" #6 0x0000000000464c32 in shutdown_children (sig=15, show_status=1) at main.c:715 __FUNCTION__ = "shutdown_children" #7 0x0000000000466223 in handle_sigs () at main.c:806 chld = 0 chld_status = 139 memlog = 0 __FUNCTION__ = "handle_sigs" #8 0x000000000046a459 in main_loop () at main.c:1767 i = 8 pid = 28926 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\326\177", '\000' <repeats 18 times>, "@\352\023/\377\177\000\000\244-\351Y\000\000\000\000\240mA\000\000\000\000\000\060\354\023/\377\177", '\000' <repeats 18 times>, "p\352\023/\377\177\000\000\030>K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #9 0x000000000046ca85 in main (argc=5, argv=0x7fff2f13ec38) at main.c:2566 cfg_stream = 0xc36010 c = -1 r = 0 tmp = 0x7fff2f13ec68 "\233\356\023/\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5c7c18 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 3628627122 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x4162c9 "H\203\304\b\303" <Address 0x4162ce out of bounds> __FUNCTION__ = "main" (gdb) Regards,

2 2

Call forwarding from external did
by arun Jayaprakash 12 Feb '14

12 Feb '14

Hello, I have set up call forwarding in Kamailio using user_preference table. When I make a call from a local extension the callfwd funcition works. The call gets forwarded to an external did number. The problem happens when and external call ( from a DID) comes to this extension the call does not get forwarded. I had to comment out the following lines in the config file to make it work: # only local users allowed to call # if((from_uri!=myself)) { # sl_send_reply("403", "Not Allowed"); # exit; # } My question is if it is a risky think to comment out these lines? If so, what are my options. Thank you, Arun

2 3

ice problems
by jaflong jaflong 12 Feb '14

12 Feb '14

I am having problems with calls from webrtc to kamailio forwarded to Asterisk These are snippet of the debug logs Asterisk CSeq: 4910 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 Jssip Cause: Bad Media Description Origin: remote Searching on google I get some indication this is to do with ice config? Please can some one suggest if this is so. In my scenerio the webrt clients will only call to the asterisk server (and not to other user agent). Considersing this I think maybe can do without ice. Is it possbile to disable ice.

3 2

About a bug when registering a Tandberg (Cisco) E20 to Kamailio - Crash Segfault at 18 IP - error 4 in auth_db.so
by Pedro Niño 12 Feb '14

12 Feb '14

Kamailio is having a crash (Segfault at 18 IP - error 4 in auth_db.so) when registering a cisco E20, via SIP Here is a full dbg backtrack, and the core dump, but have no ideas about how to proceed. Im using Kamailio 4.1.1: Got this Error at /var/log: [14629359.586980] kamailio[982]: segfault at 18 ip 00007f034db8dcfd sp 00007fff34702950 error 4 in auth_db.so[7f034db85000+c000] Kamailio Version: ---------- version: kamailio 4.1.1 (x86_64/linux) d6ffa1 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d6ffa1 compiled on 16:28:48 Jan 29 2014 with gcc 4.4.5 ------------- Thanks in Advance.

2 2

reSIProcate v1.9.0 released
by Daniel Pocock 11 Feb '14

11 Feb '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 The v1.9.0 release has been tagged (r10958) and made available for download: https://www.resiprocate.org/files/pub/reSIProcate/releases/ The release info page is here: http://www.resiprocate.org/ReSIProcate_1.9_Release and a direct link to the ChangeLog in SVN: https://svn.resiprocate.org/rep/resiprocate/tags/resiprocate-1.9.0/ChangeLog Debian, Fedora and Ubuntu packages have been updated, automated builds are in progress and they should be available shortly. Highlights of the v1.9.0 release: * Session/registration accounting * SIP over WebSocket support for WebRTC, with cookie authentication option * Python scripting for repro routes * repro dynamically-loaded routing plugins * UAS PRACK * Improvements to daemon processes for UNIX/Linux users * Android support * Multiple users in reTurn * Many stability/bug fixes SHA224 checksums for the files: resiprocate-1.9.0.tar.gz 914b01c4633e0fc03f085fdfc77199577ac079c848c35e7f04c8d344 resiprocate-contrib-1.9.0.tar.gz 444b05ae9986a43d25347e7ea4908ab3d02a1bba4d70f0744de1da11 Contributors to v1.9.x include Scott Godin, Eric Rescorla, Adam Roach, Byron Campen, Jason Fischl, Francis Joanis, Matthias Moetje, Catalin Constantin Usurelu and Daniel Pocock. We are also very grateful to all those who contributed feedback through the mailing list, testing the beta packages in Debian and all past contributors to previous releases of the reSIProcate project. Any and all feedback is welcome -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Icedove - http://www.enigmail.net/ iQIcBAEBCAAGBQJS+m+0AAoJEOm1uwJp1aqD2v4P/2/QVY5WPJ6eb7sjf/ZQ5PDs 99ed+Yy7dIRw4KqX6b4qqADesHxrvABr0mAWFTjVqJa9qWDzSINspKZyP0cq4jEi Kj1rIaGPP9cDwrlc6zAPWx+XYJSSJG/79uZYa3y6y2yV1Yupd5QM4kgCj5ZhsshN nkgtzLvpxWqyXN8ISQ415Z2kmoIQGbLpcCcKb+zQUlmyYTyUny7NB31BSGQ5Ob8N 3/t/zvyA4vee0YhoY72TfHvpdfjmDDE98x5s2UWjHg1BWDcSzthFk41TYDiPFEJt YIHPQHFKFSoyW1oa5mmsMcFbumUnhElXQsIUC1EBYUtJIl//wEL+1cZayth4GkTz klhXM5OjFs1qWn30EyGnZPMJbdAey+P2nn/EE+8XgRBxHyYI0T7zOjrcXXdBaAbX VXlRaqkb9IvpAtVuNdQWrMFBAlejIKrQs3NKTLXdW9ym3SdF8E75jr5rzILq3aWk RzAdD4HZg2ym9BIW9+jYlEmIwC5eE+C89NBCwia7K+QYf77U4Yjf89LXFUZa0L4D eGQ+aBA21qtV4bfGPf0XR+kdjqkk4erQtKxPgwIIgpDyPNnomy5nTS37v4gd0+1y hkJO0rqHSl/LeA5L2hCxkHpP8FZZ/QlmKs8niv63vfTjfldEHAAxARZG6/BDqM6H hJbfoaw7Wm3EsXj9QWYg =vu2w -----END PGP SIGNATURE-----

1 0

Kamailio World Updates
by Daniel-Constantin Mierla 11 Feb '14

11 Feb '14

From the first edition, Kamailio World tried to connect people and companies that can help each other or complement. If anyone is looking for new challenges in their career, don't hesitate to approach us. You can see more details at: - http://conference.kamailio.com/k02/2014/02/career-jobs/ Check also the Speakers <http://conference.kamailio.com/k02/speakers/> [1] and Schedule <http://conference.kamailio.com/k02/schedule/> [2] pages for new information about who is going to present and the topics. B eware of early registration period ending soon, so hurry up if you plan to attend the event – how to do it and other related details are in the Registration page <http://conference.kamailio.com/k02/registration/> [3]. Enjoy using Kamailio and looking forward to meet many of you in Berlin! Daniel/ / [1] - http://conference.kamailio.com/k02/speakers/ [2] - http://conference.kamailio.com/k02/schedule/ [3] - http://conference.kamailio.com/k02/registration/ -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Problem with Port=5061
by Owais ul Haq 10 Feb '14

10 Feb '14

Hello, I have configured TLS on kamailio server. I am done with all procedure of creation and installation of CA and site certificates. Problem is that in kamailio.cfg, when I use command port =5061. Kamailio doesnot start and error given in logs is >> ERROR: tcp_init : bind(a, 0x9007f4, 16) on MyIP:5061 : Adress already in use >> ERROR: Error while initializing TCP port of TLS Socket MyIP:5061 Point to note is that kamailio works fine with port=5062. Can somebody please help me in this regard. I will be so thankful. Cheers, Owais

4 4

Determining size of INVITE
by Daniel Tryba 10 Feb '14

10 Feb '14

How can I get the determine the total size of an outgoing INVITE? The size of the body is $bs, but is there a way to get the size of all headers? Based on the size of the total request I have to chose between UDP and TCP, upstream provider requires IP packets>1300 to be transmitted over TCP. Is there an easy way (like $bs) or should I just call msg_apply_changes() and loop through all headers and sum the lengths of them? -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 3

Get information in 'Reason' header field
by Tuan Viet Nguyen 10 Feb '14

10 Feb '14

Hello, How to get ISUP cause in the 'Reason' header field of SIP error reply ? I've tried a few ways but it did not work @message.header.Reason.params["cause"] or $hdr(Reason) gives null Thanks,

3 4

SNMP Monitoring
by Volkan Oransoy 09 Feb '14

09 Feb '14

Hi all, I have activated snmpstats module for my sip proxy. What I wonder is, which sensors are important to monitor a registrar and proxy? Which sensors do you use to monitor production servers? Thanks. /Volkan

1 0

Kamailio Admin Book
by omid mohajerani 09 Feb '14

09 Feb '14

Dear All , Would you mind tell me when is the Kamailio admin Book release time ? I want to start working with kamailio but it is very hard to start . Im sure this book will help me a lot . Regards Omid

2 2

Re: [SR-Users] Dialog
by Daniel-Constantin Mierla 09 Feb '14

09 Feb '14

Hello, do you need it in kamailio config or from an external application? From external application the mi/rpc command has to be used. Cheers, Daniel On 06/02/14 21:45, John Murray wrote: > > Hi, > > I need to get call-id and from-tag from a call in ringing state (2). > > If I use dlg_manage() and put the call into a profile using > set_dlg_profile("usr",”1"). > > Then do a avp_db_query() I don’t see the call until it is connected. > > I am using db_mode 1. > > Yet if I do a sercmd dlg.dlg_list I see the call in ringing state and > when it is connected. > > How do I get the call-id when it is in ringing state? > > Regards > > John > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 7

"Locating SIP-Servers" problem
by Franz Edler 09 Feb '14

09 Feb '14

Dear all, I have a problem to verify the correct RFC 3263 behavior of Kamailio SIP Server v4.1. I have a basic configuration of two SIP servers one domains "net1.test" and the other for domain "net2.test". When I now setup a session between both domains (from sip:alice@net1.test to sip:bob@net2.test) I expect a NAPTR-, SRV- and A-query before forwarding the INVITE to the (inbound) proxy server of net2.test. But I instead get an error: 0(3994) ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "net2.test" The trace-protocol shows only two A-queries - A-query for net2.test - A-query for net2.test.net1.test Why do I not get the NAPTR- and SRV-queries? BR Franz

2 2

Troubles with PERL module in Kamailio 3.3.5 and 4.0.3
by neumann 09 Feb '14

09 Feb '14

Hi everybody. Sorry for my bad english. Then i trying to use OpenSER module in my perl script i got the error: Can't locate object method "bootstrap" via package "OpenSER" at /usr/lib64/kamailio/perl//OpenSER/Message.pm line 33. Compilation failed in require at /usr/lib64/kamailio/perl//OpenSER.pm line 34. BEGIN failed--compilation aborted at /usr/lib64/kamailio/perl//OpenSER.pm line 34. Compilation failed in require at ./qutest.pl line 4. BEGIN failed--compilation aborted at ./qutest.pl line 4. This module is supported now? this is my script =))) ------------------------------------------ #!/usr/bin/perl use strict; use lib '/usr/lib64/kamailio/perl/'; use OpenSER; exit 0 ————————————————————— I using PERL 5.14.2 from debian wheezy repo. perv -V Summary of my perl5 (revision 5 version 14 subversion 2) configuration: Platform: osname=linux, osvers=3.2.0-4-amd64, archname=x86_64-linux-gnu-thread-multi uname='linux babin 3.2.0-4-amd64 #1 smp debian 3.2.46-1+deb7u1 x86_64 gnulinux ' config_args='-Dusethreads -Duselargefiles -Dccflags=-DDEBIAN -D_FORTIFY_SOURCE=2 -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -Dldflags= -Wl,-z,relro -Dlddlflags=-shared -Wl,-z,relro -Dcccdlflags=-fPIC -Darchname=x86_64-linux-gnu -Dprefix=/usr -Dprivlib=/usr/share/perl/5.14 -Darchlib=/usr/lib/perl/5.14 -Dvendorprefix=/usr -Dvendorlib=/usr/share/perl5 -Dvendorarch=/usr/lib/perl5 -Dsiteprefix=/usr/local -Dsitelib=/usr/local/share/perl/5.14.2 -Dsitearch=/usr/local/lib/perl/5.14.2 -Dman1dir=/usr/share/man/man1 -Dman3dir=/usr/share/man/man3 -Dsiteman1dir=/usr/local/man/man1 -Dsiteman3dir=/usr/local/man/man3 -Duse64bitint -Dman1ext=1 -Dman3ext=3perl -Dpager=/usr/bin/sensible-pager -Uafs -Ud_csh -Ud_ualarm -Uusesfio -Uusenm -Ui_libutil -DDEBUGGING=-g -Doptimize=-O2 -Duseshrplib -Dlibperl=libperl.so.5.14.2 -des' hint=recommended, useposix=true, d_sigaction=define useithreads=define, usemultiplicity=define useperlio=define, d_sfio=undef, uselargefiles=define, usesocks=undef use64bitint=define, use64bitall=define, uselongdouble=undef usemymalloc=n, bincompat5005=undef Compiler: cc='cc', ccflags ='-D_REENTRANT -D_GNU_SOURCE -DDEBIAN -fstack-protector -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64', optimize='-O2 -g', cppflags='-D_REENTRANT -D_GNU_SOURCE -DDEBIAN -fstack-protector -fno-strict-aliasing -pipe -I/usr/local/include' ccversion='', gccversion='4.7.2', gccosandvers='' intsize=4, longsize=8, ptrsize=8, doublesize=8, byteorder=12345678 d_longlong=define, longlongsize=8, d_longdbl=define, longdblsize=16 ivtype='long', ivsize=8, nvtype='double', nvsize=8, Off_t='off_t', lseeksize=8 alignbytes=8, prototype=define Linker and Libraries: ld='cc', ldflags =' -fstack-protector -L/usr/local/lib' libpth=/usr/local/lib /lib/x86_64-linux-gnu /lib/../lib /usr/lib/x86_64-linux-gnu /usr/lib/../lib /lib /usr/lib libs=-lgdbm -lgdbm_compat -ldb -ldl -lm -lpthread -lc -lcrypt perllibs=-ldl -lm -lpthread -lc -lcrypt libc=, so=so, useshrplib=true, libperl=libperl.so.5.14.2 gnulibc_version='2.13' Dynamic Linking: dlsrc=dl_dlopen.xs, dlext=so, d_dlsymun=undef, ccdlflags='-Wl,-E' cccdlflags='-fPIC', lddlflags='-shared -L/usr/local/lib -fstack-protector' Characteristics of this binary (from libperl): Compile-time options: MULTIPLICITY PERL_DONT_CREATE_GVSV PERL_IMPLICIT_CONTEXT PERL_MALLOC_WRAP PERL_PRESERVE_IVUV USE_64_BIT_ALL USE_64_BIT_INT USE_ITHREADS USE_LARGE_FILES USE_PERLIO USE_PERL_ATOF USE_REENTRANT_API Locally applied patches: DEBPKG:debian/arm_thread_stress_timeout - http://bugs.debian.org/501970 Raise the timeout of ext/threads/shared/t/stress.t to accommodate slower build hosts DEBPKG:debian/cpan_definstalldirs - Provide a sensible INSTALLDIRS default for modules installed from CPAN. DEBPKG:debian/db_file_ver - http://bugs.debian.org/340047 Remove overly restrictive DB_File version check. DEBPKG:debian/doc_info - Replace generic man(1) instructions with Debian-specific information. DEBPKG:debian/enc2xs_inc - http://bugs.debian.org/290336 Tweak enc2xs to follow symlinks and ignore missing @INC directories. DEBPKG:debian/errno_ver - http://bugs.debian.org/343351 Remove Errno version check due to upgrade problems with long-running processes. DEBPKG:debian/libperl_embed_doc - http://bugs.debian.org/186778 Note that libperl-dev package is required for embedded linking DEBPKG:fixes/respect_umask - Respect umask during installation DEBPKG:debian/writable_site_dirs - Set umask approproately for site install directories DEBPKG:debian/extutils_set_libperl_path - EU:MM: Set location of libperl.a to /usr/lib DEBPKG:debian/no_packlist_perllocal - Don't install .packlist or perllocal.pod for perl or vendor DEBPKG:debian/prefix_changes - Fiddle with *PREFIX and variables written to the makefile DEBPKG:debian/fakeroot - Postpone LD_LIBRARY_PATH evaluation to the binary targets. DEBPKG:debian/instmodsh_doc - Debian policy doesn't install .packlist files for core or vendor. DEBPKG:debian/ld_run_path - Remove standard libs from LD_RUN_PATH as per Debian policy. DEBPKG:debian/libnet_config_path - Set location of libnet.cfg to /etc/perl/Net as /usr may not be writable. DEBPKG:debian/m68k_thread_stress - http://bugs.debian.org/517938 http://bugs.debian.org/495826 Disable some threads tests on m68k for now due to missing TLS. DEBPKG:debian/mod_paths - Tweak @INC ordering for Debian DEBPKG:debian/module_build_man_extensions - http://bugs.debian.org/479460 Adjust Module::Build manual page extensions for the Debian Perl policy DEBPKG:debian/prune_libs - http://bugs.debian.org/128355 Prune the list of libraries wanted to what we actually need. DEBPKG:fixes/net_smtp_docs - [rt.cpan.org #36038] http://bugs.debian.org/100195 Document the Net::SMTP 'Port' option DEBPKG:debian/perlivp - http://bugs.debian.org/510895 Make perlivp skip include directories in /usr/local DEBPKG:debian/cpanplus_definstalldirs - http://bugs.debian.org/533707 Configure CPANPLUS to use the site directories by default. DEBPKG:debian/cpanplus_config_path - Save local versions of CPANPLUS::Config::System into /etc/perl. DEBPKG:debian/deprecate-with-apt - http://bugs.debian.org/580034 Point users to Debian packages of deprecated core modules DEBPKG:fixes/hurd-ccflags - [a190e64] http://bugs.debian.org/587901 [perl #92244] Make hints/gnu.sh append to $ccflags rather than overriding them DEBPKG:debian/squelch-locale-warnings - http://bugs.debian.org/508764 Squelch locale warnings in Debian package maintainer scripts DEBPKG:debian/skip-upstream-git-tests - Skip tests specific to the upstream Git repository DEBPKG:fixes/extutils-cbuilder-cflags - [011e8fb] http://bugs.debian.org/624460 [perl #89478] Append CFLAGS and LDFLAGS to their Config.pm counterparts in EU::CBuilder DEBPKG:fixes/module-build-home-directory - http://bugs.debian.org/624850 [rt.cpan.org #67893] Fix failing tilde test when run under a UID without a passwd entry DEBPKG:debian/patchlevel - http://bugs.debian.org/567489 List packaged patches for 5.14.2-21+deb7u1 in patchlevel.h DEBPKG:fixes/h2ph-multiarch - [e7ec705] http://bugs.debian.org/625808 [perl #90122] Make h2ph correctly search gcc include directories DEBPKG:fixes/index-tainting - [3b36395] http://bugs.debian.org/291450 [perl #64804] RT 64804: tainting with index() of a constant DEBPKG:fixes/document_makemaker_ccflags - http://bugs.debian.org/628522 [rt.cpan.org #68613] Document that CCFLAGS should include $Config{ccflags} DEBPKG:fixes/sys-syslog-socket-timeout-kfreebsd.patch - http://bugs.debian.org/627821 [rt.cpan.org #69997] Use a socket timeout on GNU/kFreeBSD to catch ICMP port unreachable messages DEBPKG:fixes/hurd-hints - http://bugs.debian.org/636609 Improve general GNU hints, needed for GNU/Hurd. DEBPKG:fixes/pod_fixes - [7698aed] http://bugs.debian.org/637816 Fix typos in several pod/perl*.pod files DEBPKG:debian/find_html2text - http://bugs.debian.org/640479 Configure CPAN::Distribution with correct name of html2text DEBPKG:fixes/digest_eval_hole - http://bugs.debian.org/644108 Close the eval "require $module" security hole in Digest->new($algorithm) DEBPKG:fixes/hurd-ndbm - [f0d0a20] [perl #102680] http://bugs.debian.org/645989 Add GNU/Hurd hints for NDBM_File DEBPKG:fixes/sysconf.t-posix - [8040185] [perl #102888] http://bugs.debian.org/646016 Fix hang in ext/POSIX/t/sysconf.t on GNU/Hurd DEBPKG:fixes/hurd-largefile - [1fda587] [perl #103014] http://bugs.debian.org/645790 enable LFS on GNU/Hurd DEBPKG:debian/hurd_test_todo_syslog - http://bugs.debian.org/650093 Disable failing GNU/Hurd tests in cpan/Sys-Syslog/t/syslog.t DEBPKG:fixes/hurd_skip_itimer_virtual - [rt.cpan.org #72754] http://bugs.debian.org/650094 Skip interval timer tests in Time::HiRes on GNU/Hurd DEBPKG:debian/hurd_test_skip_sigdispatch - http://bugs.debian.org/650188 Disable failing GNU/Hurd tests op/sigdispatch.t DEBPKG:debian/hurd_test_skip_stack - http://bugs.debian.org/650175 Disable failing GNU/Hurd tests dist/threads/t/stack.t DEBPKG:debian/hurd_test_skip_pipe - http://bugs.debian.org/650187 Disable failing GNU/Hurd tests io/pipe.t DEBPKG:debian/hurd_test_skip_io_pipe - http://bugs.debian.org/650096 Disable failing GNU/Hurd tests dist/IO/t/io_pipe.t DEBPKG:fixes/manpage_name_CPAN - http://bugs.debian.org/650448 [rt.cpan.org #73396] cpan/CPAN: add NAME headings in modules with POD DEBPKG:fixes/manpage_name_CPANPLUS - http://bugs.debian.org/650450 [rt.cpan.org #73398] cpan/CPANPLUS: add NAME headings in modules with POD DEBPKG:fixes/manpage_name_Test-Harness - http://bugs.debian.org/650451 [rt.cpan.org #73399] cpan/Test-Harness: add NAME headings in modules with POD DEBPKG:fixes/manpage_name_Term-UI - http://bugs.debian.org/650452 [rt.cpan.org #73400] cpan/Term-UI: add NAME headings in modules with POD DEBPKG:fixes/podlators_ae_ligature_fallback - http://bugs.debian.org/652851 Fix the ASCII fallback string for AE DEBPKG:fixes/fsf_postal_address - [de89470] Update references to the FSF's postal address DEBPKG:fixes/cpan_module_pod_fixes - [perl #106870] [rt.cpan.org #73447] [rt.cpan.org #73446] Fix POD formatting in Term-Cap and Pod-Parser DEBPKG:fixes/cgi_no_shellwords_pl - Use Text::ParseWords instead of shellwords.pl DEBPKG:fixes/path_max_fallback - [perl #109262] http://bugs.debian.org/656869 Don't use _POSIX_PATH_MAX as a fallback PATH_MAX DEBPKG:debian/makemaker-pasthru - http://bugs.debian.org/660195 [rt.cpan.org #28632] Make EU::MM pass LD through to recursive Makefile.PL invocations DEBPKG:fixes/propagate_tainted_errors.patch - http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=663158 [perl #111654] properly propagate tainted errors DEBPKG:debian/perl5db-x-terminal-emulator.patch - http://bugs.debian.org/668490 Invoke x-terminal-emulator rather than xterm in perl5db.pl DEBPKG:fixes/socket_cache_propagate - http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=659075 [rt.cpan.org #61577] [perl #112736] sockdomain and socktype undef on newly accepted sockets DEBPKG:fixes/ipc_open3 - [perl #114454] http://bugs.debian.org/683894 IPC::Open3::open3(..., '-') broken DEBPKG:fixes/string_repeat_overrun - http://bugs.debian.org/689314 [b675304] avoid calling memset with a negative count DEBPKG:debian/cpan-missing-site-dirs - http://bugs.debian.org/688842 Fix CPAN::FirstTime defaults with nonexisting site dirs if a parent is writable DEBPKG:fixes/kfreebsd-overrides - http://bugs.debian.org/689713 [perl #115324] [7dc6565] Remove unnecessary overrides in gnukfreebsd and gnuknetbsd hints. DEBPKG:fixes/tainted-smartmatch - [be88a5c] http://bugs.debian.org/690571 [perl #93590] $tainted ~~ [...] failing DEBPKG:fixes/regexp-matching-starter - [6e634c5] http://bugs.debian.org/690975 [perl #101710] Regression with /i, latin1 chars. DEBPKG:fixes/regexp-matching-fold - [399fb9c] http://bugs.debian.org/690976 regexec.c: Fix "\x{FB01}\x{FB00}" =~ /ff/i DEBPKG:fixes/regexp-matching-opposite-case - [dc91d5a] http://bugs.debian.org/690979 [perl #101970] /[[:lower:]]/i matches upper case DEBPKG:fixes/reading-glob-copy-handle - [fd1564b] http://bugs.debian.org/629363 [perl #92258] <$fh> hangs on a glob copy DEBPKG:fixes/smartmatch-rhs-precedence - http://bugs.debian.org/691102 [011be0b] Enforce Any ~~ Object smartmatch precedence DEBPKG:fixes/perlcheat-update - http://bugs.debian.org/691112 [ab0ae0a] Update PerlCheat to 5.14 DEBPKG:fixes/cgi-cr-escaping - http://bugs.debian.org/693420 CR escaping for P3P and Set-Cookie headers DEBPKG:fixes/maketext-code-execution - [1735f6f] http://bugs.debian.org/695224 Fix misparsing of maketext strings. DEBPKG:fixes/storable-security-warning - [664f237] http://bugs.debian.org/695223 add a note about security concerns in Storable DEBPKG:fixes/digest-sha-doublefree - [rt.cpan.org #82655] http://bugs.debian.org/698172 [a8c6ff7] Fix a double-free bug in Digest::SHA DEBPKG:fixes/64bitint-signedness-wraparound - http://bugs.debian.org/698320 [94e529c] Avoid wraparound when casting unsigned size_t to signed ssize_t. DEBPKG:fixes/stdin-sigchld - http://bugs.debian.org/700171 [perl #116621] [be48bbe] add a couple missing LEAVEs in perlio_async_run() DEBPKG:fixes/hsplit-rehash - [d59e31f] http://bugs.debian.org/702296 Prevent premature hsplit() calls, and only trigger REHASH after hsplit() DEBPKG:fixes/encode-memleak - http://bugs.debian.org/702416 [5814803] Encode: Fixed a memory leak that occurred in the UTF-8 encoding. DEBPKG:fixes/threads_shared_elements_crash - [perl #119089] http://bugs.debian.org/718438 threads::shared should not crash if shared elements outlive their aggregate. DEBPKG:fixes/perlbug-patchlist - [3541c11] http://bugs.debian.org/710842 [perl #118433] Make perlbug look up the list of local patches at run time DEBPKG:fixes/digest_sha_double_free - [ee8c6f4] [rt.cpan.org #86295] http://bugs.debian.org/711206 maint-5.18: Digest-SHA crash fix in 5.85 DEBPKG:fixes/pl_eval_start_use_after_free - [eae139f] [perl #115992] PL_eval_start use-after-free DEBPKG:fixes/regcomp_fix_segv - [ebb390a] [perl #115994] fix segv in regcomp.c:S_join_exact() DEBPKG:fixes/list_util_off_by_two - [623a911] fix off-by-two error in List::Util DEBPKG:fixes/sdbm_off_by_one - [7f5f08b] [perl #111586] sdbm.c: fix off-by-one access to global ".dir" DEBPKG:fixes/socket_unpack_sockaddr_un_heap_buffer_overflow - [e508642] [perl #111594] Socket::unpack_sockaddr_un heap-buffer-overflow Built under linux Compiled at Sep 30 2013 03:45:34 @INC: /etc/perl /usr/local/lib/perl/5.14.2 /usr/local/share/perl/5.14.2 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.14 /usr/share/perl/5.14 /usr/local/lib/site_perl . ———————————— Timofeev Dmitry VoIP Engineer Linux, Asterisk, Cisco solutions Skype: itsroot icq: 227227933

2 1

Increasing MAX_RECV_BUFFER_SIZE
by Wingsravi R 09 Feb '14

09 Feb '14

Dear Kamailio'ns I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on Ubuntu 12.04. I am experiencing some Latency, jitter, pixelled, choppy video errors in my set-up. So i wanted to increase buffer sizes of my kamailio server , Mediaproxy server and Ubuntu also. In that regards i have increased Shared memory value to 512 MB and Package memory value 16 MB. BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the value for 'MAX_RECV_BUFFER_SIZE' of kamailio? And are there any settings to increase Buffer Size of Mediaproxy server ? Is this changes will affect on my issues ? PS: I even changed ubuntu settings also : net.core.rmem_max= 8388608 net.core.wmem_max= 8388608 (but these values reset to default values again when i restarted system.) Please help me in resolving this issues. Any help will appreciate. Regards, Ravi

4 4

kamailo uses wrong ip address
by jaflong jaflong 07 Feb '14

07 Feb '14

I have this in my config $ru = "sip:"+$rU+"10.31.2.101:5090;transport=tls"; if (!t_relay_to_tls()) { sl_reply_error(); return; } My ip address dest is 10.31.2.101. Why has 130.31.2.101 been used? 13(10882) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 130.31.2.101 4(10883) ERROR: <core> [tcp_main.c:4422]: tcpconn_main_timeout(): connect 130.31.2.101:5090 failed (timeout)

2 1

My Kamailio + Asterisk != NOT realtime integration: TRANSFERERNAME and BLINDTRANSFER
by Alexandr Usov 07 Feb '14

07 Feb '14

Integration - works. Problem - dialing peer to peer via Kamailio OK but with missing VARs and extension number, on dialing/transferring. Maybe you know other way to configure Asterisk dialplan for users, registered on kamailio and alowing dial as SIP/UserNumber insted SIP/ UserNumber(a)kamailio.host.name What we have. My Kamailio config: *http://pastebin.com/p7YxsFaw <http://pastebin.com/p7YxsFaw>* Asterisk's user (peer) - registered on kamailio: [3] host=192.168.144.212 qualify=yes dtmfmode=rfc2833 canreinvite=no context=local-routing host=dynamic type=friend directmedia=no nat=no qualify=yes disallow=all allow=ulaw allow=alaw allow=g729 call-limit=2 limitonpeers=yes callcounter=yes callerid=Usov Mob <3> ### Asterisk queue members: ;syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse] member=SIP/1(a)sip.cloudpbx.com.ua,1,1001,SIP/1,no member=SIP/3(a)sip.cloudpbx.com.ua,1,1003,SIP/1,no ### Asterisk dialing local peer: ... exten => _X,n,Dial(SIP/${EXTEN}(a)sip.cloudpbx.com.ua,12,tT) ... ### Asterisk attended transfer from 1 to 9 exten: *http://pastebin.com/k9H4vMgx <http://pastebin.com/k9H4vMgx>* Problem #1: Peer 9 receive clid as asterisk(a)sip.cloudpbx.com.ua Need #1: 1(a)sip.cloudpbx.com.ua. Problem #2: TRANSFERERNAME=SIP/sip.cloudpbx.com.ua-000000b5 Need #2: TRANSFERERNAME=SIP/1(a)sip.cloudpbx.com.ua-000000b5 ### Asterisk blind transfer dump log: *http://pastebin.com/NXqXingR <http://pastebin.com/NXqXingR>* Problem #3: BLINDTRANSFER=SIP/sip.cloudpbx.com.ua-000000ba Need #3: BLINDTRANSFER=SIP/9(a)sip.cloudpbx.com.ua-000000ba Main problem it is missed peer number - SIP/*peer_number*@ sip.cloudpbx.com.ua. And asterisk(a)sip.cloudpbx.com.ua instead 1(a)sip.cloudpbx.com.ua on attended transfer. Asterisk sip.conf domain settings: realm=sip.cloudpbx.com.ua fromdomain=sip.cloudpbx.com.ua ;domain=sip.cloudpbx.com.ua ;; temporary not uses becaouse not accepting GOIP sim ports registration on asterisk with IP-address of sip proxy instead domain sip.cloudpbx.com.ua; and it's not helps to change asterisk(a)sip.cloudpbx.com.ua clid Any help from all - wellcome! P.S. domain sip.cloudpbx.com.ua not exist

1 1

CONFIRMED_NA dialog state timeout
by Alex Balashov 07 Feb '14

07 Feb '14

Hi, We frequently run into the situation where a call is simultaneously CANCEL'd by the caller and answered (2xx) by the callee. This results in the caller not sending an e2e ACK, since it CANCEL'd the branch. It results in retransmission of the 200 OK, since the proxy can't CANCEL the branch anymore. It leaves the dialog in state 3 #define DLG_STATE_CONFIRMED_NA 3 /*!< confirmed dialog without a ACK yet */ (dlg_hash.h) for up to 'dialog_timeout' time. Is there a straightforward hack for this situation? Does changing the dialog timeout AVP have any effect after the relay of the initial INVITE (and thus invocation of the registered dialog init TM callback)? More generally, I was wondering if there has been any thought given to the idea of separate timeout modparams for confirmed-nonACKed state, as well as early state perhaps. Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

ssl error
by jaflong jaflong 06 Feb '14

06 Feb '14

This is my tls.cfg for server [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/asterisk/certs/proxy.key certificate = /etc/asterisk/certs/proxy.crt As far as I understand (verify_certificate = no), and (require_certificate = no) should allow a client connecting without certicates. Can anyone understand what this debug indicates What is causes this error tls_read_f(): TLS read:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca 7(14841) DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> 7(14841) DEBUG: tls [tls_domain.c:609]: sr_ssl_ctx_info_callback(): SSL handshake started 7(14841) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 7(14841) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7eff24e41260 n=940 fd=5 7(14841) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 7(14841) DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8df1e0, 5, 2, 0x7eff24e41260), fd_no=1 7(14841) DEBUG: tls [tls_domain.c:621]: sr_ssl_ctx_info_callback(): SSL handshake done 7(14841) DEBUG: tls [tls_domain.c:624]: sr_ssl_ctx_info_callback(): SSL disable renegotiation 7(14841) DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful 7(14841) DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 10.1.2.3:35573 using TLSv1/SSLv3 AES256-SHA 256 7(14841) DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 10.1.2.4:6443 7(14841) DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate 7(14841) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 7(14841) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7eff24e41260 n=250 fd=5 7(14841) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 7(14841) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7eff24e41260, FD 5 7(14841) ERROR: tls [tls_server.c:1186]: tls_read_f(): TLS read:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca 7(14841) ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading 7(14841) DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8df1e0, 5, -1, 0x10) fd_no=2 called 7(14841) DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7eff24e41260, state -2, fd=5, id=2 7(14841) DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7eff24e3c200 10(14844) DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7eff24e41260, -2 from 1 10(14844) DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7eff24e3c200

2 1

Dialog
by John Murray 06 Feb '14

06 Feb '14

Hi, I need to get call-id and from-tag from a call in ringing state (2). If I use dlg_manage() and put the call into a profile using set_dlg_profile("usr","1"). Then do a avp_db_query() I don't see the call until it is connected. I am using db_mode 1. Yet if I do a sercmd dlg.dlg_list I see the call in ringing state and when it is connected. How do I get the call-id when it is in ringing state? Regards John

1 0

Change of source ip address
by Diego Alejandro Ozuna Escalada 06 Feb '14

06 Feb '14

Dear list, I'm having a problem. I have an SBC server with two network interfaces, one external IP (X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the proxy sip server through internal ip with this configuration: route [sip-serverr] { rewritehostport ("ip_sip_proxy: 5060"); force_send_socket (X.X.248.21: 5060); t_relay (); exit; } When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct external source IP (X.X.248.194) but if there are retransmissions of these responses, after 6 seconds, the source IP changes to the internal ip (X.X.248.21) Does anyone from the list know how I can solve this? Thanks, Regards, -- Diego

2 2

FS#380 - fixes for memory leak in TLS module
by Ding Ma 06 Feb '14

06 Feb '14

Daniel, Klaus, Sorry to reply late to your comments, was busy doing some other things until now. Just provided some comments in https://sip-router.org/tracker/index.php?do=details&task_id=380 Please take a look and let me know if you agree with my assessment. Thanks, Ding

1 0

bug in authdb_mod ??
by jay binks 06 Feb '14

06 Feb '14

So when Im trying to use is_subscriber & load_credentials together, It seems I have either hit a bug or just a bad config ( but I dont think its the latter ) My symptom is that only the field on the far right of "load_credentials" ever returns the correct value, the other fields are always null. My config has this setting : modparam("auth_db", "load_credentials", "field1;field2") and I can see the following debug when running kamailio -ddddd 0(14174) DEBUG: auth_db [authdb_mod.c:291]: w_is_subscriber(): uri [sip: 1112(a)192.168.173.5] table [subscriber] flags [1] 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:807]: db_cassa_query(): query table=subscriber 0(14174) DEBUG: db_cassandra [dbcassa_table.c:461]: dbcassa_db_get_table(): found dbname=kamailio, table=subscriber 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:351]: cassa_constr_key(): query col = username 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:351]: cassa_constr_key(): query col = domain 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:376]: cassa_constr_key(): key = 1112 192.168.173.5 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:499]: cassa_translate_query(): Query col: field2 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:499]: cassa_translate_query(): Query col: field1; field2 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:501]: cassa_translate_query(): get 2 columns 0(14174) DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7fc6ce770e78 if I had to put money on where the bug is, I would look in parse_aaa_pvs in authdb_mod.c however, an initial glance revealed nothing glaring, also this file was last changed in *2013-01-20 ( and that was only a move from modules_k )* SURELY I cant be the only person wanting to use this for more than one field ? This code looks like its almost the same as when first committed in August 2011 .... Id love someone to see if they can confirm this and point me in the right direction ( if this is my fault ) or lets get this fixed, if its a bug :) -- Sincerely Jay

2 5

Handling RFC3578 (Overlap Dialing) in Kamailio/ Asynchronious transaction handling
by Moritz Graf 06 Feb '14

06 Feb '14

Hoho, i want to implement RFC3578 (transaction oriented) in Kamailio. Therefore I have to correlate two different Invites with same Call-ID. I see various ways of accomplishing this, but I am stuck at all of my solutions. Find the algorithm I want to implement attached. Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) The following lines, I'm going to explain my thoughts. For reference I'm calling "Thread A" the first INVITE coming in and "Thread B" the second. ####################### ### using Module ASYNC * Thread A -> async_route("handle_overlap","<timer t99 value>"); * Thread B -> if new incoming transaction, ** set avp("<call-id>")= <digitlength of called number> and ** call async_route("handle_overlap","<timer t99 value>"); * in route[handle_overlap] ** if avp("call-id") greater than "<digitlength of called number>" *** cancel transaction with "484 -Address Incomplete" ** if avp("call-id") equals "<digitlength of called number>" *** handle call => the number is complete PROBLEM: The old transaction is not canceled immediately, but after t99 elapsed... ####################### ### using Module TMX using t_suspend() and t_continue(), but here i see the following problems: * Thread A calls "t_suspend()" ** How is it possible to cancel "Thread A"'s transaction from Thread B? ** If there is no Thread B, so the numbers were complete, how can I get Thread A to continue processing?? * General TMX-Question: Are there any examples available? ####################### ### using simple sleep() and shv()/avp() Sounds like dirty hacking... I'd appriciate a solution with one of aboves modules. But if there is a solution available, I'd try this. Thx in advise Moritz -- Moritz Graf, B.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg Telefon +49 (9 41) 69 85 - 1 86 Telefax +49 (9 41) 69 85 - 2 86 mailto:moritz.graf@g-fit.de http://www.g-fit.de G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co. KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg, HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher

4 11

Kamailio + CDR-stats
by davy 06 Feb '14

06 Feb '14

Hi, I’ve recently tried out CDR-stats, and after a bumpy ride to get it going ;) , I’m happily surprised with it. Especially the fraud detection mechanisms (which could use some extra’s), and the eye-candy makes it attractive. It is based on a mongodb which has to be filled with CDRs. As there is no mongodb module in Kamailio, I was thinking of using the jsonrpc module to push cdrs from Kamailio to some Python thing which pushes it into MongoDB. Perhaps by using the async module. Netstrings seems a rather low overhead protocol, and the jsonrpc module allows to push over a cluster of servers, which makes it scalable. Would this be a good way to do this, or am I bluntly missing the better way? Thx, Davy

3 2

dialog module - dialogs not deleted on BYE
by jay binks 06 Feb '14

06 Feb '14

Im fairly use to the dialog module, and I can not seem to get the dialogs to be removed from the dialogs table on hangup ( BYE ). What am I missing here.. this seems very basic and is essential to using dialog module for for channel limiting. any help would be great. maybe there is an example config for this somewhere ? -- Sincerely Jay

2 3

MI commands ?
by arun Jayaprakash 06 Feb '14

06 Feb '14

Hello, what do I need to do to be able to run MI commands on Kamailio 4.0? When I run MI commands from Siremis no results come back. Thank you. Arun

2 4

kamailio as sip router and rtp proxy without reg on kamailio
by *sm1Ly 05 Feb '14

05 Feb '14

hello all again. trying the third day... my cfg is: http://pastebin.com/vMHd7kki I need scheme like: any_sip_device_on_internet_nated_and_not => kamailio ext ip (194.190.8.171) | kamailio int ip (50.0.0.1) => asterisk (50.0.0.10) a do not need realtime configuration of asterisk. I only need forward everything to asterisk. and after signalling will work, I need rtp too... but its the long way I think. so how I must config kamailio that it doesnt answer 200 when I send register? it looks like sip device send register, kamailio fwd it, but answer to device 200. and than my device doesnt resend register normal. some dump: 20:40:44.649660 IP (tos 0x0, ttl 121, id 14950, offset 0, flags [none], proto UDP (17), length 583) 77.37.241.151.5060 > 194.190.8.171.5060: SIP, length: 555 REGISTER sip:194.190.8.171 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b95a985a85ad1a82;rport From: <sip:1001@194.190.8.171>;tag=3041452802 To: <sip:1001@194.190.8.171> Call-ID: 00C48A3A-288C-E311-B956-985A85AD1A82(a)192.168.1.119 CSeq: 4 REGISTER Contact: <sip:1001@192.168.1.119:5060 >;Expires=0;+sip.instance="<urn:uuid:8012D316-1188-E311-99FF-536675A7CDE6>" Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 User-Agent: SIPPER for PhonerLite Expires: 0 Content-Length: 0 20:40:44.651668 IP (tos 0x10, ttl 64, id 33739, offset 0, flags [none], proto UDP (17), length 422) 194.190.8.171.5060 > 77.37.241.151.5060: SIP, length: 394 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b95a985a85ad1a82;rport=5060;received=77.37.241.151 From: <sip:1001@194.190.8.171>;tag=3041452802 To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.63a1 Call-ID: 00C48A3A-288C-E311-B956-985A85AD1A82(a)192.168.1.119 CSeq: 4 REGISTER Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 20:40:44.653916 IP (tos 0x10, ttl 64, id 33740, offset 0, flags [none], proto UDP (17), length 582) 194.190.8.171.5060 > 77.37.241.151.5060: SIP, length: 554 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK009318ba288ce311b95a985a85ad1a82;rport=5060 From: <sip:1001@194.190.8.171>;tag=3041452802 To: <sip:1001@194.190.8.171>;tag=as40c64a63 Call-ID: 00C48A3A-288C-E311-B956-985A85AD1A82(a)192.168.1.119 CSeq: 4 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cffbc95" Content-Length: 0 20:40:45.073854 IP (tos 0x0, ttl 121, id 14953, offset 0, flags [none], proto UDP (17), length 575) 77.37.241.151.5060 > 194.190.8.171.5060: SIP, length: 547 REGISTER sip:194.190.8.171 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b1713a64da149374;rport From: <sip:1001@194.190.8.171>;tag=3123771917 To: <sip:1001@194.190.8.171> Call-ID: 009318BA-288C-E311-B16F-3A64DA149374(a)192.168.1.119 CSeq: 1 REGISTER Contact: <sip:1001@192.168.1.119:5060 >;+sip.instance="<urn:uuid:8012D316-1188-E311-99FF-536675A7CDE6>" Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 User-Agent: SIPPER for PhonerLite Expires: 900 Content-Length: 0 20:40:45.076169 IP (tos 0x10, ttl 64, id 33741, offset 0, flags [none], proto UDP (17), length 572) 194.190.8.171.5060 > 77.37.241.151.5060: SIP, length: 544 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b1713a64da149374;rport=5060;received=77.37.241.151 From: <sip:1001@194.190.8.171>;tag=3123771917 To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.795d Call-ID: 009318BA-288C-E311-B16F-3A64DA149374(a)192.168.1.119 CSeq: 1 REGISTER Contact: <sip:1001@192.168.1.119:5060>;expires=900;received="sip: 77.37.241.151:5060 ";+sip.instance="<urn:uuid:8012D316-1188-E311-99FF-536675A7CDE6>" Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 -- yours respectfully, Alexander Vasin. 8 926 1437200 icq: 9906064

2 1

NAT helper frustrations with multiple kamailios
by Daniel Tryba 05 Feb '14

05 Feb '14

I'm experiencing some problems getting 2 Kamailios to work together. A clients - A kamailio - B kamailio - B clients | the outside world A and B client may or may not be NATed to their own kamailio instance. A and B kamailio (with forced rtpproxies) are in the same public network. A-kamailio and its clients are essentially a reseller of my services on the B side. The problem arises when A or B incorrectly flag a message as needing NAT help so either an unnecessary add_contact_alias is added (which seems to happen when putting somebody on hold on an A-client). A-kamailio inserts an alias on the Contact header containing the ip/port of B-kamailio. Trying to resume the call sends the contact with the alias to B-kamailio, which does a handle_ruri_alias and thus sets $du to itself, the party that was on hold will never receive these messages. The problem I have with add_contact_alias/handle_ruri_alias is that any kamailio in the path will parse these hints whether or not it is actually ment for that kamailio. I'd rather see a scheme that somehow identifies if the added alias was generated locally and thus needs to be handled or can just be ignored. Anybody else got these problems/frustrations and maybe a hint for an other way to tackle this problem? Off to figure out why uac_test incorrectly flags the onhold process as needing the nat helper... -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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Re: [SR-Users] Kamailio not mdifying Contact in multi-homed (*sm1Ly)
by *sm1Ly 05 Feb '14

05 Feb '14

I got really modify my config, and now it looks like: http://pastebin.com/0M7938Mm but now I got loopback problems. kamailio just asking itself. ngrep -d lo port 5060 talks me: U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;branch=z9hG4bKc2bb.e627f45ee16a4b45ca99321b3bc8832c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.328c8c23df8e916010525f7f786cbc49.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060 ;branch=z9hG4bKc2bb.b8ec8f173657be182ad714f59d37d4a8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..V ia: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2 bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 1 94.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a 7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rpor t=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988 c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x 86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.328c8c23df8e916010525f7f786cbc49.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b8ec8f173657be182ad714f59d37d4a8.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b 3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01d cad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190 .8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001 @194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b8ec8f173657be182ad714f59d37d4a8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf 4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946 cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168 .1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757D AE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c 09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0ea f0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..Fro m: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length : 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b 43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d 28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba4655 7f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0 c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branc h=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: < sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE ..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e 520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=12744772 58..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf7 3e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SI P/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558 B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c 7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14df b;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: < sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux) )..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66 b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171 >;tag=c5a ea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173 b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77 .37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB(a)192.168.1.119.. CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc28 04cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8. 171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d 2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call -ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be3 11988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1 .1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.1 90.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119 :5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DF B@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <si p:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0... . and how looks dump 1. from sip device to kamailio [root@kamaz sm1ly]# tcpdump -nn -i eth0 host 77.37.241.151 -v tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 19:54:51.205942 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 421) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 393 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Content-Length: 0 19:54:51.210060 IP (tos 0x10, ttl 64, id 35861, offset 0, flags [none], proto UDP (17), length 603) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 575 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as518dbea9 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31daf0f8" Content-Length: 0 19:54:51.257552 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 586) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 558 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="31daf0f8",uri="sip: 194.190.8.171:5060 ",response="2b0222b07da7af1c3b72c2eb36241f5f",algorithm=MD5 Content-Length: 0 19:54:51.261209 IP (tos 0x10, ttl 64, id 35862, offset 0, flags [none], proto UDP (17), length 603) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 575 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f70c759" Content-Length: 0 19:54:51.290458 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 586) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 558 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="2f70c759",uri="sip: 194.190.8.171:5060 ",response="9eb7d157cfd0d8e8575b59c94c5f82f6",algorithm=MD5 Content-Length: 0 19:54:51.295285 IP (tos 0x10, ttl 64, id 35863, offset 0, flags [none], proto UDP (17), length 566) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 538 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.314824 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 473) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 445 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Content-Length: 0 19:54:51.319697 IP (tos 0x10, ttl 64, id 35864, offset 0, flags [none], proto UDP (17), length 602) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 574 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as0bba2324 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="023deb34" Content-Length: 0 19:54:51.344080 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 638) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 610 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="023deb34",uri="sip: 194.190.8.171:5060 ",response="7fa5be98c7c0c395ed41f9e7da095629",algorithm=MD5 Content-Length: 0 19:54:51.350423 IP (tos 0x10, ttl 64, id 35865, offset 0, flags [none], proto UDP (17), length 602) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 574 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26886c3d" Content-Length: 0 19:54:51.370976 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 638) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 610 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="26886c3d",uri="sip: 194.190.8.171:5060 ",response="cc17aae74d45f06e4712aa66b08e68df",algorithm=MD5 Content-Length: 0 19:54:51.380962 IP (tos 0x10, ttl 64, id 35866, offset 0, flags [none], proto UDP (17), length 636) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 608 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:1002@192.168.1.103:57288;transport=udp>;expires=3600 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.406160 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 395) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 367 OPTIONS sip:194.190.8.171 SIP/2.0 Call-ID: 28b382a24bc3e426f030593cec5f852a(a)192.168.1.103 CSeq: 1287 OPTIONS From: "1002" <sip:1002@194.190.8.171>;tag=3866552602 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK9bddae72a5ccaf0df80be00ba2882dfd353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Content-Length: 0 19:54:51.407799 IP (tos 0x10, ttl 64, id 35867, offset 0, flags [none], proto UDP (17), length 458) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 430 SIP/2.0 484 Address Incomplete Call-ID: 28b382a24bc3e426f030593cec5f852a(a)192.168.1.103 CSeq: 1287 OPTIONS From: "1002" <sip:1002@194.190.8.171>;tag=3866552602 To: "1002" <sip:1002@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.5162 Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK9bddae72a5ccaf0df80be00ba2882dfd353236;rport=57288;received=77.37.241.151 Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 2. from kamailio to asterisk. [root@kamaz sm1ly]# tcpdump -nn -i eth1 host 50.0.0.10 -v and port 5060 tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 65535 bytes 19:54:41.906391 IP (tos 0x0, ttl 64, id 5618, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:42.906385 IP (tos 0x0, ttl 64, id 5619, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:43.906565 IP (tos 0x0, ttl 64, id 5620, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:44.904726 IP (tos 0x10, ttl 64, id 34743, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-8578 Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 19:54:51.207422 IP (tos 0x10, ttl 64, id 34744, offset 0, flags [none], proto UDP (17), length 531) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 503 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Content-Length: 0 19:54:51.209052 IP (tos 0x0, ttl 64, id 5621, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as518dbea9 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31daf0f8" Content-Length: 0 19:54:51.258759 IP (tos 0x10, ttl 64, id 34745, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="31daf0f8",uri="sip: 194.190.8.171:5060 ",response="2b0222b07da7af1c3b72c2eb36241f5f",algorithm=MD5 Content-Length: 0 19:54:51.259902 IP (tos 0x0, ttl 64, id 5622, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f70c759" Content-Length: 0 19:54:51.292190 IP (tos 0x10, ttl 64, id 34746, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="2f70c759",uri="sip: 194.190.8.171:5060 ",response="9eb7d157cfd0d8e8575b59c94c5f82f6",algorithm=MD5 Content-Length: 0 19:54:51.293302 IP (tos 0x0, ttl 64, id 5623, offset 0, flags [none], proto UDP (17), length 676) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 648 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.317242 IP (tos 0x10, ttl 64, id 34747, offset 0, flags [none], proto UDP (17), length 583) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 555 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Content-Length: 0 19:54:51.318123 IP (tos 0x0, ttl 64, id 5624, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as0bba2324 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="023deb34" Content-Length: 0 19:54:51.346522 IP (tos 0x10, ttl 64, id 34748, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="023deb34",uri="sip: 194.190.8.171:5060 ",response="7fa5be98c7c0c395ed41f9e7da095629",algorithm=MD5 Content-Length: 0 19:54:51.348056 IP (tos 0x0, ttl 64, id 5625, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26886c3d" Content-Length: 0 19:54:51.373298 IP (tos 0x10, ttl 64, id 34749, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="26886c3d",uri="sip: 194.190.8.171:5060 ",response="cc17aae74d45f06e4712aa66b08e68df",algorithm=MD5 Content-Length: 0 19:54:51.374893 IP (tos 0x0, ttl 64, id 5626, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:51.374906 IP (tos 0x0, ttl 64, id 5627, offset 0, flags [none], proto UDP (17), length 746) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 718 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:1002@192.168.1.103:57288;transport=udp>;expires=3600 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:52.375165 IP (tos 0x0, ttl 64, id 5628, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:53.375333 IP (tos 0x0, ttl 64, id 5629, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:54.342025 IP (tos 0x10, ttl 64, id 34750, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-1f6e Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 3. on asterisk to kamailio [root@astkamaz sm1ly]# tcpdump -nn -i eth0 -v host 50.0.0.1 and port 5060 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 19:54:41.906401 IP (tos 0x0, ttl 64, id 5618, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:42.906581 IP (tos 0x0, ttl 64, id 5619, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:43.906740 IP (tos 0x0, ttl 64, id 5620, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:44.905366 IP (tos 0x10, ttl 64, id 34743, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-8578 Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 19:54:51.208464 IP (tos 0x10, ttl 64, id 34744, offset 0, flags [none], proto UDP (17), length 531) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 503 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Content-Length: 0 19:54:51.209176 IP (tos 0x0, ttl 64, id 5621, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as518dbea9 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31daf0f8" Content-Length: 0 19:54:51.259436 IP (tos 0x10, ttl 64, id 34745, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="31daf0f8",uri="sip: 194.190.8.171:5060 ",response="2b0222b07da7af1c3b72c2eb36241f5f",algorithm=MD5 Content-Length: 0 19:54:51.260101 IP (tos 0x0, ttl 64, id 5622, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f70c759" Content-Length: 0 19:54:51.292708 IP (tos 0x10, ttl 64, id 34746, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="2f70c759",uri="sip: 194.190.8.171:5060 ",response="9eb7d157cfd0d8e8575b59c94c5f82f6",algorithm=MD5 Content-Length: 0 19:54:51.293541 IP (tos 0x0, ttl 64, id 5623, offset 0, flags [none], proto UDP (17), length 676) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 648 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.317704 IP (tos 0x10, ttl 64, id 34747, offset 0, flags [none], proto UDP (17), length 583) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 555 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Content-Length: 0 19:54:51.318355 IP (tos 0x0, ttl 64, id 5624, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as0bba2324 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="023deb34" Content-Length: 0 19:54:51.347429 IP (tos 0x10, ttl 64, id 34748, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="023deb34",uri="sip: 194.190.8.171:5060 ",response="7fa5be98c7c0c395ed41f9e7da095629",algorithm=MD5 Content-Length: 0 19:54:51.348212 IP (tos 0x0, ttl 64, id 5625, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26886c3d" Content-Length: 0 19:54:51.373777 IP (tos 0x10, ttl 64, id 34749, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="26886c3d",uri="sip: 194.190.8.171:5060 ",response="cc17aae74d45f06e4712aa66b08e68df",algorithm=MD5 Content-Length: 0 19:54:51.374731 IP (tos 0x0, ttl 64, id 5626, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:51.374804 IP (tos 0x0, ttl 64, id 5627, offset 0, flags [none], proto UDP (17), length 746) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 718 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:1002@192.168.1.103:57288;transport=udp>;expires=3600 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:52.375348 IP (tos 0x0, ttl 64, id 5628, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:53.375562 IP (tos 0x0, ttl 64, id 5629, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:54.342525 IP (tos 0x10, ttl 64, id 34750, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-1f6e Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 ####### Defined Values ######### # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### # Enable debugging only when kamailio is mading boo-boo ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=3 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=8 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ #alias="kamaz.ikatele.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:50.0.0.1:5060 advertise 194.190.8.171:5060 #listen=udp:194.190.8.171:5060 advertise 50.0.0.1:5060 listen=udp:194.190.8.171 listen=udp:50.0.0.1 dns=no rev_dns=no check_via=yes /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 mhomed=1 ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/lib64/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "nathelper.so" loadmodule "rtpproxy.so" #loadmodule "topoh.so" #modparam("topoh", "mask_key", "balalayka") #modparam("topoh", "mask_ip", "50.0.0.1") # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 3000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 12000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@194.190.8.171") # params needed for NAT traversal in other modules modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } route { # per request initial checks route(REQINIT); # NAT detection xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); route(NATDETECT); xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in Route[NATDETECT] fix_nat-register\n"); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(RELAY); } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("20")) { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MAX FDW HEADERS ALARMO\n"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route route[NATDETECT] { force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } return; } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(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 route(RELAY); exit; } else { xlog("L_NOTICE","ACK without matching transaction ... ignore and discard\n"); # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { # if(isflagset(FLT_NATS)) # { # setbflag(FLB_NATB); # # uncomment next line to do SIP NAT pinging # xlog("L_NOTICE","--------------------- SIP NAT PINGING --------------------------------------\n"); # setbflag(FLB_NATSIPPING); # } # if (!save("location")) # sl_reply_error(); # xlog("L_NOTICE","--------------------- SL REPLY ERROR --------------------------------------\n"); if( t_check_trans() ) { xlog("L_NOTICE","Transaction already exist, forwarding\n"); } else { xlog("L_NOTICE","New transaction upcoming, creating\n"); # forward("50.0.0.10"); # save("sip:50.0.0.10","0x02"); t_relay_to_udp("50.0.0.10", "5060"); exit; } exit; } } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_NOTICE","Default routing block raised!\n"); if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_NOTICE","------------------------------------ SETBFLAG ----------------------------\n"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_NOTICE","------------------------------------ ADD RR PARAM NAT YES ----------------------------\n"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) xlog("L_NOTICE","------------------------------------ IS FIRST HOP SRT CONACT ALIAS ----------------------------\n"); set_contact_alias(); } xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[NATMANAGE] RTPproxy with EI Flags\n"); } return; } # URI update for dialog requests route[DLGURI] { if(!isdsturiset()) { handle_ruri_alias(); xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[DLGURI] RTPproxy with EI Flags\n"); } return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MANAGE_BRANCH\n"); xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MANAGE_REPLY\n"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MANAGE_FAILURE\n"); route(NATMANAGE); if (t_is_canceled()) { exit; } } route[DEBUG] { if (method==("PUBLISH|SUBSCRIBE|REGISTER|OPTIONS")) { return; } xlog("[$mi] Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); } > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 3 Feb 2014 15:14:19 +0400 > From: "*sm1Ly" <st.uzver(a)gmail.com> > To: dipak.biswas(a)mezocliq.com, sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Kamailio not mdifying Contact in multi-homed > setup > Message-ID: > < > CABCivXiED_CDQiJtgQ-p_XQ1h8p3rbkE3fnxnS2iuUH+kdHZvA(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > hello, I got the same issue. my sip device getting ok from kamailio, but > asterisk reply 401. > but. > this is my config: http://pastebin.com/jGCak01E > > and I changed regfwd route to > $du = "sip:50.0.0.10:5060;transport=udp"; > forward(); > > and now I see this in logs: > > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:637]: parse_msg(): status: <200> > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:639]: parse_msg(): reason: <OK> > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, > <branch> = <0>; state=16 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, > flags=2 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the > first via > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [receive.c:152]: receive_msg(): After parse_msg... > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm > [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global > id=24 T start=(nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: > tag=006cfccc318be31188fc19977b3a5651 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, > state=29 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> > [63]; uri=[sip:77.37.241.151:5060] > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip: > 77.37.241.151:5060>] > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> > <OPTIONS> > *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm > [t_lookup.c:1045]: t_reply_matching(): DEBUG: t_reply_matching: failure to > match a transaction* > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm > [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global > id=25 T end=(nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : > content_length=0 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:106]: get_hdr_field(): found end of header > *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [forward.c:784]: do_forward_reply(): reply cannot be forwarded - no 2nd > via* > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [receive.c:296]: receive_msg(): receive_msg: cleaning up > > > any ideas? > and show me ur conf please. > > > -- > > Hi All, > > I'm running into an issue, I'm not sure whether any of you seen this > yourselves and resolved it. Please share some pointers. My network is: > > clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 > (Freeswitch) > > The 200 OK response from Freeswitch (on the way back from called party to > caller) to Kamailio is shown below. Notice the Contact header URI host part > contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change > that to Public IP before forwarding the 200 OK (copied below) to Caller in > public domain. But, it's not. As a result, ACK from Caller is not reaching > back to Kamailio. > > How did you or anybody out there using Kamailio resolve this problem? If > needed, I can copy/paste my kamailio.cfg. > > -- > yours respectfully, Alexander Vasin. > > 8 926 1437200 > icq: 9906064 >

2 1

Re: [SR-Users] reg event support in subscribe
by Andrew Pogrebennyk 05 Feb '14

05 Feb '14

Hi, See presence_reginfo - Extension to Presence server for registration info replication (RFC3680) http://kamailio.org/docs/modules/stable/modules/presence_reginfo.html On 02/05/2014 07:56 AM, Premchandiran wrote: > Hi All, > > May I know whether kamailio supports event header with reg (EVENT:reg) > (rfc 3680) , if kamailio supports may I know the module whether it is in > presence or presence_xml or any other module? > > > > Regards, > > *Prem Chandiran M*** >

1 0

Problem with response of http_query of the moudule utils
by Aman Agarwal 05 Feb '14

05 Feb '14

Hi, I am using http_query to make some HTTP API calls. I have found that the response of the HTTP call always has the initial 1062 characters missing. The HTTP call is sending strings of different lengths but the initial 1062 characters are always missing. Can anybody help me with this issue or has any idea why this might be happening? Thanks Aman Agarwal

1 0

Msilo message body type
by sipatse＠in.gr 05 Feb '14

05 Feb '14

1 0

websocket relay
by jaflong jaflong 05 Feb '14

05 Feb '14

Hi Everyone, I want kamailio to do a single task, that is relay incoming (webrtc sip over websockets) connections to asterisk. All sip handling is to be done is asterisk. I have configured asterisk with tls settings. My sample conf file is below At the moment I am getting these errors ERROR: tm [ut.h:343]: uri2dst2(): no corresponding socket for af 2 ERROR: tm [t_fwd.c:520]: prepare_new_uac(): ERROR: can't fwd to af 2, proto 3 (no corresponding listening socket) Please can anyone provide any advise to complete this configuration listen=tcp:10.2.3.4:5080 enable_tls=yes tcp_connection_lifetime=3605 tcp_accept_no_cl=yes ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/local/lib64/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "sdpops.so" loadmodule "nathelper.so" loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") ####### Routing Logic ######## event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm =~"GET") { if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not Found", "", ""); } request_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } route(REQINIT); route(NATDETECT); route(RELAY); } route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } route[NATDETECT] { if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) fix_nated_register(); else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } } route[RELAY] { xlog("L_INFO", "Forwarding Request\n"); $ru = "sip:"+$rU+"10.2.3.5:5080;transport=tcp"; if (!t_relay_to_tls()) { sl_reply_error(); return; } exit; }

2 1

reg event support in subscribe
by Premchandiran 05 Feb '14

05 Feb '14

Hi All, May I know whether kamailio supports event header with reg (EVENT:reg) (rfc 3680) , if kamailio supports may I know the module whether it is in presence or presence_xml or any other module? Regards, Prem Chandiran M

1 0

GSM PDU Decoding
by Ramya Ramamurthy 04 Feb '14

04 Feb '14

Dear SR-Users, I am using kamailio for decoding of SIP Packets with Message Body as a GSM PDU. But I am not sure of the modules which i need to use for the same. Some help on this would be greatly appreciated. Thanks, ~Ramya.

5 7

Change the name of the location table for the usrloc module
by Alejandro Recarey 04 Feb '14

04 Feb '14

Hi everybody, I have been trying to setup a new installation of kamailio, and I need the user location table to have a specific name. This will allow two kamailio servers to use the same database for their registry information. I cannot find any modparam in the documentation that allows me to set the table name for the usrloc. Is there any way to do this? It seems like an oversight, every other module I have been able to change the table name without a problem. Thank you, Alex

2 2

Cluster: check/mark servers in cluster
by Victor V. Kustov 04 Feb '14

04 Feb '14

Hi, all! I had in config: #!define KAMDB "cluster://postcash" loadmodule "db_cluster.so" modparam("db_cluster", "connection","con1=>postgres://user1:password1@host1/kamailio") modparam("db_cluster", "connection","con2=>postgres://user2:password2@host2/kamailio") modparam("db_cluster", "cluster", "postcash=>con1=9s8p;con2=9s8p") modparam("usrloc", "db_url", KAMDB) and save registered user's locations in cluster after some server's restarts second base has empty set. How can I see server's list with status? How can I set second server worked? (I check - all OK with it) -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.12.9-calculate GNU/Linux

1 0

Hello from newbie
by Alex Lake 04 Feb '14

04 Feb '14

Just trying to get to grips with Kamailio and thought I'd say hello to you all... Not sure if this is of any interest, but I've found it quite hard to get into Kamailio compared to (say) Freeswitch. This is not meant as criticism at all, but if you're looking to make it easier, I may be able to help! Installation was more or less OK, although a couple of slightly blind alleys and lack of concrete examples of config files. I believe I have a working server, but where to go next has left me a little stumped! What I'm trying to do is have a SIP load-balancer such that I can have a set of Freeswitch servers that can have traffic routed to them on a controlled basis, allowing me to point certain accounts at certain servers and take individual servers out of service for upgrade etc. I think this is SIP server redirection. If there was an example of how to do it - eg. what files need to be changed, any database entries, etc. that would be most useful. Unfortunately, all my Googling/list searching so far has only come up with questions that were either badly worded or not answered. Happy to turn anything I come up with into a beginner-friendly case-study.

2 5

Asterisk re-INVITE race condition, error 500.
by David K 04 Feb '14

04 Feb '14

Hello all, So I have three machines, we don't care about audio for this problem, so everything I mention here is SIP related. Freeswitch <--> Kamailio 3.3.2 <--> Asterisk 1. Asterisk sends an INVITE to Freeswitch through the Kamailio proxy. 2. Kamailio replies 100 Trying and forwards to Freeswitch 3. Freeswitch replies 100 Trying 4. Freeswitch replies 180 Ringing to Kamailio 5. Kamailio routes the answer to Asterisk 6. Freeswitch replies 200 OK to Kamailio 7. Kamailio replies 200 OK to Asterisk 8. Asterisk replies ACK to Kamailio 9. Asterisk sends a re-INVITE to Freeswitch through Kamailio 10. Kamailio routes the re-INVITE to freeswitch 11. Kamailio routes the ACK to freeswitch. 12. Freeswitch replies 500 Server error because it got a re-INVITE before the ACK. So, my problem is that Kamailio seems to process my re-INVITE more quickly than the ACK. So Freeswitch replies an error because it got the re-INVITE before the ACK. So my "solution" is to add a usleep(20); for re-INVITEs on Kamailio, but I think this is a lousy solution. Has anyone here had to deal with problems where Kamailio routes a re-INVITE faster than an ACK causing endpoints to return error messages? Has anyone had to deal with a similar issue? Thanks, David

6 11

Kamailio World 2014 - New speakers and presentations
by Daniel-Constantin Mierla 04 Feb '14

04 Feb '14

Slightly less than two months to the start of Kamailio World 2014 <http://www.kamailioworld.com> [1] the organization of the event is getting accelerated. We received plenty of interesting proposals so far, we expect to end the Call for Speakers very soon, earlier than previously announced. If you have plans to submit, you should really hurry up. The profiles of about 20 speakers have been published in Speakers page <http://conference.kamailio.com/k02/speakers/> [2], few others that were accepted already will be added soon. Details for about 15 sessions during the conference days (April 3-4, 2014) are available in the Schedule page <http://conference.kamailio.com/k02/schedule/> [3]. The technical workshop on the 2nd of April is getting to the final shape as well. Similar to last year, expect several exhibition spots where various companies will present their services or products during the conference days. In the evening of 3rd of April it will be the cocktail party, a relaxing event for social networking around good food and drinks. We are grateful to our sponsors [4], they are making possible this event: FhG Fokus <http://www.fokus.fraunhofer.de/en/fokus/index.html>, Asipto <http://www.asipto.com>, Sipwise <http://www.sipwise.com>, Sipgate <http://www.sipgate.de>, IT Center <http://www.itcenter.com.pt/> and NG Voice <http://www.ng-voice.com/>. Should you consider sponsoring, contact us via email at <conference [at] kamailio.com>**and we can present you more details. The event take place again at the very city center of Berlin, next to Island of Museums and Alexanderplatz. The number of seats is limited and they can fill up very quickly, if you plan to attend, do not delay the registration – see more details in Registration page <http://conference.kamailio.com/k02/registration/> [5]. /Looking forward to meeting many of you in Berlin! / Daniel/ / [1] - http://conference.kamailio.com [2] - http://conference.kamailio.com/k02/speakers/ [3] - http://conference.kamailio.com/k02/schedule/ [4] - http://conference.kamailio.com/k02/sponsors/ [5] - http://conference.kamailio.com/k02/registration/ -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Proxied authentication
by Pete Ashdown 04 Feb '14

04 Feb '14

I'm having trouble divining the proper way to authenticate calls to remote destinations that require it. I'm hoping someone can clear it up. Freeswitch client server A is authenticated to Kamailio. Freeswitch test server B is authenticated to Kamailio. Calls from A -> Kamailio -> B fail with rejection of the authentication information that A is using to Kamailio. As far as I understand, I need to rewrite the authentication information with UAC? However, UAC's uacreg table seems to tie remote users to remote destinations, when all I want to do is authenticate all calls routed to the destination regardless of what the user or the source is. Am I using the wrong module?

2 1

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Shankar 04 Feb '14

04 Feb '14

Hi, Can someone clarify here please? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, February 03, 2014 10:53 AM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Regd. MAX_LDG_LOCKS Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Regards, Shankar

1 0

Media-proxy Query
by Nandini madhu 04 Feb '14

04 Feb '14

Dear All, Greetings, I have configured Kamailio (4.0.3) with mediaproxy server, with this set-up audio/video calls are going good and hence NAT issue has solved. I have used two soft clients: 1) IMSDroid 2) Bria. But i have a query about media-proxy statistics i.e in the syslog, the info about media-proxy activity is shown as below: *Case 1*: when Call between two IMSDroid clients. caller_codec : 'Unknown(73)' callee_codec : 'Unknown(73)' callee_ua : 'unknown agent'. *Case 2* : When call between two Bria clients. 'callee_ua': 'unknown agent'. (Calls using Bria clients media-proxy's syslog shows Audio codec names, but using IMSdroid clients it is showing 'Unknown' for Audio codec names ). Find the attachment below for the full syslog about the mentioned unknown behaviour. And also find my kamailio.cfg file. And also logging in the syslog as shown like RTP: Unknown , RTCP: Unknown. In this context, there is a jitter in audio calls and pixelled video sometimes, So is there any wrong with the above mentioned "unknown' factor in this problem of jitter/pixelled audio/video calls ? (AFAIK these payload type 73 (i.e 72-76) is reserved for RTCP conflict avoidance. right? ) What could be the problem for this 'Unknown' issue? How can i solve this issue ? Anything can be done in kamailio script? And how these RTCP payload types (72-76) plays a role in audio/video calls? Any suggestions will greatly help. please help me in clarifying these issues. Thanks in advance. Regards, Nandini

2 3

Buffer Size
by Wingsravi R 04 Feb '14

04 Feb '14

Dear Kamailio'ns I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on Ubuntu 12.04. I am experiencing Latency, jitter, pixel led, choppy video errors in my set-up. So i wanted increase buffer sizes in my kamailio server , Mediaproxy server and Ubuntu also. In that regards i have increased Shared memory value to 512 MB and Package memory value 16 MB. BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the value for 'MAX_RECV_BUFFER_SIZE' ? And are there any settings to increase Buffer Size of Mediaproxy server ? Is this changes will affect on my issues ? PS: I even changed ubuntu settings also : net.core.rmem_max= 8388608 net.core.wmem_max= 8388608 (but these values reset to default values again when i restarted system.) Please help me in resolving this issues. Any help will appreciate. Regards, Ravi

1 0

Re: [SR-Users] dialog list query
by John Murray 03 Feb '14

03 Feb '14

From: Charles Chance <charles.chance(a)sipcentric.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] dialog list query Message-ID: Charles hi, >What db_mode are you using for dialog module? Mode 5. I Don't need to persist the dialog state so means table maintained in memory and not flushed to disk until shutdown but I can access it through normal avp_db_query right? >Are the entries in the db if you check manually? If I do sercmd dlg.dlg_list I see the call fine including callid etc. >Are you able to fetch other data using avp_db_query? Yes lots of queries to non dialog tables working fine. >And do you see any errors in the log? No logs show query happening fine and returning Null. Thanks John

2 4

Re: [SR-Users] Kamailio not mdifying Contact in multi-homed setup
by *sm1Ly 03 Feb '14

03 Feb '14

hello, I got the same issue. my sip device getting ok from kamailio, but asterisk reply 401. but. this is my config: http://pastebin.com/jGCak01E and I changed regfwd route to $du = "sip:50.0.0.10:5060;transport=udp"; forward(); and now I see this in logs: Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <200> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <OK> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <0>; state=16 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global id=24 T start=(nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=006cfccc318be31188fc19977b3a5651 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, state=29 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [63]; uri=[sip:77.37.241.151:5060] Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip: 77.37.241.151:5060>] Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <OPTIONS> *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1045]: t_reply_matching(): DEBUG: t_reply_matching: failure to match a transaction* Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global id=25 T end=(nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [forward.c:784]: do_forward_reply(): reply cannot be forwarded - no 2nd via* Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [receive.c:296]: receive_msg(): receive_msg: cleaning up any ideas? and show me ur conf please. -- Hi All, I'm running into an issue, I'm not sure whether any of you seen this yourselves and resolved it. Please share some pointers. My network is: clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch) The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But, it's not. As a result, ACK from Caller is not reaching back to Kamailio. How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg. -- yours respectfully, Alexander Vasin. 8 926 1437200 icq: 9906064

2 1

Change of source ip address
by Diego Alejandro Ozuna Escalada 03 Feb '14

03 Feb '14

Dear list, I'm having a problem. I have an SBC server with two network interfaces, one external IP (X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the proxy sip server through internal ip with this configuration: route [sip-serverr] { rewritehostport ("ip_sip_proxy: 5060"); force_send_socket (X.X.248.21: 5060); t_relay (); exit; } When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct external source IP (X.X.248.194) but if there are retransmissions of these responses, after 6 seconds, the source IP changes to the internal ip (X.X.248.21) Does anyone from the list know how I can solve this? Attached picture of wireshark capture for better understanding. -- Diego Alejandro Ozuna

1 0

Security hygiene for Kamailio
by davy 03 Feb '14

03 Feb '14

Hi all, we all enjoy our FAIL2BAN and snippets of our Kamailio config when we see it successfully fight off the "friendly-scanner", and multiple futile attempts to fool our systems. But it got me thinking… What is a sufficient level of security on our Kamailio machinery… ? Are we all just doing whatever, or is the nature of the beast, that every setup is different? Eventually while having a beer, we will end up in the discussion Kamailio is as good (and even much better) as most of the commercially available SBCs. But, imho, that all depends on the configuration. There are a few good reads available, and on the security front I personally love Pike, Topoh, Dnssec, Htable and recently I think I'm doing rather clever stuff with CNXCC… And I do feel comfortable on my setups, them won't be hacked… But do we have a-sort -of stake in the ground example configuration which we can consider as being more than sufficiently secure? Some config where we can tick off all the known security risks for SIP (as chapter 26 of rfc3261 gives a state of the art back in 2002) Or would that be a nice idea for a micro project? Grtz, Davy

6 12

Carrierroute - scan_prefix
by Oliver Roth 03 Feb '14

03 Feb '14

Hi all I need a solution to get the used scan_prefix in carrierroute module. We have different carriers for our destinations, sometimes split up 50%/50% or other splittings - all routes with failure routes. Now if one destination is not reachable, then I would like to get this information and reroute to another carrier. As we use the carrierroute module, we use the "description" to get the carriername (for billing purposes). Is there a solution to get the used scan_prefix from the carrierroute module? Regards, Oli

2 2

kamailio answers instead asterisk
by *sm1Ly 03 Feb '14

03 Feb '14

hello all. this is my kamailio conf: http://pastebin.com/jGCak01E when I try to register any sip device. it looks like. device -> ext IP kamailio -> int IP kamailio -> asterisk. so I trying to nat something to lan. but on asterisk I see that register, it answer 401, than register again and it again answer 401. but on my sip device I see register and than kamailio answer 200 !! why ??? 1. #!KAMAILIO 2. # 3. # Kamailio (OpenSER) SIP Server v4.1 - default configuration script 4. # - web: http://www.kamailio.org 5. # - git: http://sip-router.org 6. # 7. # Direct your questions about this file to: < sr-users(a)lists.sip-router.org> 8. # 9. # Refer to the Core CookBook at http://www.kamailio.org/wiki/ 10. # for an explanation of possible statements, functions and parameters. 11. # 12. # Several features can be enabled using '#!define WITH_FEATURE' directives: 13. # 14. # *** To run in debug mode: 15. # - define WITH_DEBUG 16. # 17. # *** To enable mysql: 18. # - define WITH_MYSQL 19. # 20. # *** To enable authentication execute: 21. # - enable mysql 22. # - define WITH_AUTH 23. # - add users using 'kamctl' 24. # 25. # *** To enable IP authentication execute: 26. # - enable mysql 27. # - enable authentication 28. # - define WITH_IPAUTH 29. # - add IP addresses with group id '1' to 'address' table 30. # 31. # *** To enable persistent user location execute: 32. # - enable mysql 33. # - define WITH_USRLOCDB 34. # 35. # *** To enable presence server execute: 36. # - enable mysql 37. # - define WITH_PRESENCE 38. # 39. # *** To enable nat traversal execute: 40. # - define WITH_NAT 41. # - install RTPProxy: http://www.rtpproxy.org 42. # - start RTPProxy: 43. # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 44. # 45. # *** To enable PSTN gateway routing execute: 46. # - define WITH_PSTN 47. # - set the value of pstn.gw_ip 48. # - check route[PSTN] for regexp routing condition 49. # 50. # *** To enable database aliases lookup execute: 51. # - enable mysql 52. # - define WITH_ALIASDB 53. # 54. # *** To enable speed dial lookup execute: 55. # - enable mysql 56. # - define WITH_SPEEDDIAL 57. # 58. # *** To enable multi-domain support execute: 59. # - enable mysql 60. # - define WITH_MULTIDOMAIN 61. # 62. # *** To enable TLS support execute: 63. # - adjust CFGDIR/tls.cfg as needed 64. # - define WITH_TLS 65. # 66. # *** To enable XMLRPC support execute: 67. # - define WITH_XMLRPC 68. # - adjust route[XMLRPC] for access policy 69. # 70. # *** To enable anti-flood detection execute: 71. # - adjust pike and htable=>ipban settings as needed (default is 72. # block if more than 16 requests in 2 seconds and ban for 300 seconds) 73. # - define WITH_ANTIFLOOD 74. # 75. # *** To block 3XX redirect replies execute: 76. # - define WITH_BLOCK3XX 77. # 78. # *** To enable VoiceMail routing execute: 79. # - define WITH_VOICEMAIL 80. # - set the value of voicemail.srv_ip 81. # - adjust the value of voicemail.srv_port 82. # 83. # *** To enhance accounting execute: 84. # - enable mysql 85. # - define WITH_ACCDB 86. # - add following columns to database 87. #!ifdef ACCDB_COMMENT 88. ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; 89. ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; 90. ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; 91. ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; 92. ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; 93. ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; 94. ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; 95. ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; 96. ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; 97. ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; 98. ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; 99. ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; 100. #!endif 101. 102. # #!define WITH_MYSQL 103. # #!define WITH_AUTH 104. # #!define WITH_USRLOCDB 105. #!define WITH_NAT 106. #!define WITH_ASTERISK 107. # #!define WITH_DEBUG 108. 109. ####### Include Local Config If Exists ######### 110. import_file "kamailio-local.cfg" 111. 112. ####### Defined Values ######### 113. 114. # *** Value defines - IDs used later in config 115. #!ifdef WITH_MYSQL 116. # - database URL - used to connect to database server by modules such 117. # as: auth_db, acc, usrloc, a.s.o. 118. #!ifndef DBURL 119. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" 120. #!endif 121. #!endif 122. #!ifdef WITH_MULTIDOMAIN 123. # - the value for 'use_domain' parameters 124. #!define MULTIDOMAIN 1 125. #!else 126. #!define MULTIDOMAIN 0 127. #!endif 128. 129. # - flags 130. # FLT_ - per transaction (message) flags 131. # FLB_ - per branch flags 132. #!define FLT_ACC 1 133. #!define FLT_ACCMISSED 2 134. #!define FLT_ACCFAILED 3 135. #!define FLT_NATS 5 136. 137. #!define FLB_NATB 6 138. #!define FLB_NATSIPPING 7 139. 140. ####### Global Parameters ######### 141. 142. ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR 143. #!ifdef WITH_DEBUG 144. debug=4 145. log_stderror=yes 146. #!else 147. debug=3 148. log_stderror=no 149. #!endif 150. 151. memdbg=5 152. memlog=5 153. 154. log_facility=LOG_LOCAL0 155. 156. fork=yes 157. children=4 158. 159. /* uncomment the next line to disable TCP (default on) */ 160. #disable_tcp=yes 161. 162. /* uncomment the next line to disable the auto discovery of local aliases 163. based on reverse DNS on IPs (default on) */ 164. #auto_aliases=no 165. 166. /* add local domain aliases */ 167. #alias="sip.mydomain.com" 168. 169. /* uncomment and configure the following line if you want Kamailio to 170. bind on a specific interface/port/proto (default bind on all available) */ 171. 172. #listen=udp:50.0.0.1:5060 advertise 194.190.8.171:5060 173. listen=udp:194.190.8.171:5060 advertise 50.0.0.1:5060 174. 175. /* port to listen to 176. * - can be specified more than once if needed to listen on many ports */ 177. port=5060 178. 179. mhomed=1 180. 181. #!ifdef WITH_TLS 182. enable_tls=yes 183. #!endif 184. 185. # life time of TCP connection when there is no traffic 186. # - a bit higher than registration expires to cope with UA behind NAT 187. tcp_connection_lifetime=3605 188. 189. ####### Custom Parameters ######### 190. 191. # These parameters can be modified runtime via RPC interface 192. # - see the documentation of 'cfg_rpc' module. 193. # 194. # Format: group.id = value 'desc' description 195. # Access: $sel(cfg_get.group.id) or @cfg_get.group.id 196. # 197. 198. #!ifdef WITH_PSTN 199. # PSTN GW Routing 200. # 201. # - pstn.gw_ip: valid IP or hostname as string value, example: 202. # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" 203. # 204. # - by default is empty to avoid misrouting 205. #pstn.gw_ip = "" desc "PSTN GW Address" 206. #pstn.gw_port = "" desc "PSTN GW Port" 207. #!endif 208. 209. #!ifdef WITH_VOICEMAIL 210. # VoiceMail Routing on offline, busy or no answer 211. # 212. # - by default Voicemail server IP is empty to avoid misrouting 213. #voicemail.srv_ip = "" desc "VoiceMail IP Address" 214. #voicemail.srv_port = "5060" desc "VoiceMail Port" 215. #!endif 216. 217. #!ifdef WITH_ASTERISK 218. asterisk.bindip = "50.0.0.10" desc "Asterisk IP Address" 219. asterisk.bindport = "5060" desc "Asterisk Port" 220. kamailio.bindip = "50.0.0.1" desc "Kamailio IP Address" 221. kamailio.bindport = "5060" desc "Kamailio Port" 222. #!endif 223. 224. ####### Modules Section ######## 225. 226. # set paths to location of modules (to sources or installation folders) 227. #!ifdef WITH_SRCPATH 228. mpath="modules/" 229. #!else 230. mpath="/usr/lib64/kamailio/modules/" 231. #!endif 232. 233. #!ifdef WITH_MYSQL 234. loadmodule "db_mysql.so" 235. #!endif 236. 237. loadmodule "mi_fifo.so" 238. loadmodule "kex.so" 239. loadmodule "tm.so" 240. loadmodule "tmx.so" 241. loadmodule "sl.so" 242. loadmodule "rr.so" 243. loadmodule "pv.so" 244. loadmodule "maxfwd.so" 245. loadmodule "usrloc.so" 246. loadmodule "registrar.so" 247. loadmodule "textops.so" 248. loadmodule "siputils.so" 249. loadmodule "xlog.so" 250. loadmodule "sanity.so" 251. loadmodule "ctl.so" 252. loadmodule "cfg_rpc.so" 253. loadmodule "mi_rpc.so" 254. loadmodule "acc.so" 255. 256. #!ifdef WITH_AUTH 257. loadmodule "auth.so" 258. loadmodule "auth_db.so" 259. #!ifdef WITH_IPAUTH 260. loadmodule "permissions.so" 261. #!endif 262. #!endif 263. 264. #!ifdef WITH_ALIASDB 265. loadmodule "alias_db.so" 266. #!endif 267. 268. #!ifdef WITH_SPEEDDIAL 269. loadmodule "speeddial.so" 270. #!endif 271. 272. #!ifdef WITH_MULTIDOMAIN 273. loadmodule "domain.so" 274. #!endif 275. 276. #!ifdef WITH_PRESENCE 277. loadmodule "presence.so" 278. loadmodule "presence_xml.so" 279. #!endif 280. 281. #!ifdef WITH_NAT 282. loadmodule "nathelper.so" 283. loadmodule "rtpproxy.so" 284. #!endif 285. 286. #!ifdef WITH_TLS 287. loadmodule "tls.so" 288. #!endif 289. 290. #!ifdef WITH_ANTIFLOOD 291. loadmodule "htable.so" 292. loadmodule "pike.so" 293. #!endif 294. 295. #!ifdef WITH_XMLRPC 296. loadmodule "xmlrpc.so" 297. #!endif 298. 299. #!ifdef WITH_DEBUG 300. loadmodule "debugger.so" 301. #!endif 302. 303. #!ifdef WITH_ASTERISK 304. loadmodule "uac.so" 305. #!endif 306. 307. 308. #loadmodule "topoh.so" 309. #modparam("topoh", "mask_key", "balalayka") 310. #modparam("topoh", "mask_ip", "50.0.0.1") 311. 312. # ----------------- setting module-specific parameters --------------- 313. 314. 315. # ----- mi_fifo params ----- 316. modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") 317. 318. 319. # ----- tm params ----- 320. # auto-discard branches from previous serial forking leg 321. modparam("tm", "failure_reply_mode", 3) 322. # default retransmission timeout: 30sec 323. modparam("tm", "fr_timer", 30000) 324. # default invite retransmission timeout after 1xx: 120sec 325. modparam("tm", "fr_inv_timer", 120000) 326. 327. 328. # ----- rr params ----- 329. # add value to ;lr param to cope with most of the UAs 330. modparam("rr", "enable_full_lr", 1) 331. # do not append from tag to the RR (no need for this script) 332. #!ifdef WITH_ASTERISK 333. modparam("rr", "append_fromtag", 1) 334. #!else 335. modparam("rr", "append_fromtag", 0) 336. #!endif 337. 338. 339. # ----- registrar params ----- 340. modparam("registrar", "method_filtering", 1) 341. /* uncomment the next line to disable parallel forking via location */ 342. # modparam("registrar", "append_branches", 0) 343. /* uncomment the next line not to allow more than 10 contacts per AOR */ 344. #modparam("registrar", "max_contacts", 10) 345. # max value for expires of registrations 346. modparam("registrar", "max_expires", 3600) 347. # set it to 1 to enable GRUU 348. modparam("registrar", "gruu_enabled", 0) 349. 350. 351. # ----- acc params ----- 352. /* what special events should be accounted ? */ 353. modparam("acc", "early_media", 0) 354. modparam("acc", "report_ack", 0) 355. modparam("acc", "report_cancels", 0) 356. /* by default ww do not adjust the direct of the sequential requests. 357. if you enable this parameter, be sure the enable "append_fromtag" 358. in "rr" module */ 359. modparam("acc", "detect_direction", 0) 360. /* account triggers (flags) */ 361. modparam("acc", "log_flag", FLT_ACC) 362. modparam("acc", "log_missed_flag", FLT_ACCMISSED) 363. modparam("acc", "log_extra", 364. "src_user=$fU;src_domain=$fd;src_ip=$si;" 365. "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") 366. modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) 367. /* enhanced DB accounting */ 368. #!ifdef WITH_ACCDB 369. modparam("acc", "db_flag", FLT_ACC) 370. modparam("acc", "db_missed_flag", FLT_ACCMISSED) 371. modparam("acc", "db_url", DBURL) 372. modparam("acc", "db_extra", 373. "src_user=$fU;src_domain=$fd;src_ip=$si;" 374. "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") 375. #!endif 376. 377. 378. # ----- usrloc params ----- 379. /* enable DB persistency for location entries */ 380. #!ifdef WITH_USRLOCDB 381. modparam("usrloc", "db_url", DBURL) 382. modparam("usrloc", "db_mode", 2) 383. modparam("usrloc", "use_domain", MULTIDOMAIN) 384. #!endif 385. 386. 387. # ----- auth_db params ----- 388. #!ifdef WITH_AUTH 389. modparam("auth_db", "db_url", DBURL) 390. modparam("auth_db", "calculate_ha1", yes) 391. modparam("auth_db", "password_column", "password") 392. modparam("auth_db", "load_credentials", "") 393. modparam("auth_db", "use_domain", MULTIDOMAIN) 394. 395. # ----- permissions params ----- 396. #!ifdef WITH_IPAUTH 397. modparam("permissions", "db_url", DBURL) 398. modparam("permissions", "db_mode", 1) 399. #!endif 400. 401. #!endif 402. 403. 404. # ----- alias_db params ----- 405. #!ifdef WITH_ALIASDB 406. modparam("alias_db", "db_url", DBURL) 407. modparam("alias_db", "use_domain", MULTIDOMAIN) 408. #!endif 409. 410. 411. # ----- speeddial params ----- 412. #!ifdef WITH_SPEEDDIAL 413. modparam("speeddial", "db_url", DBURL) 414. modparam("speeddial", "use_domain", MULTIDOMAIN) 415. #!endif 416. 417. 418. # ----- domain params ----- 419. #!ifdef WITH_MULTIDOMAIN 420. modparam("domain", "db_url", DBURL) 421. # register callback to match myself condition with domains list 422. modparam("domain", "register_myself", 1) 423. #!endif 424. 425. 426. #!ifdef WITH_PRESENCE 427. # ----- presence params ----- 428. modparam("presence", "db_url", DBURL) 429. 430. # ----- presence_xml params ----- 431. modparam("presence_xml", "db_url", DBURL) 432. modparam("presence_xml", "force_active", 1) 433. #!endif 434. 435. 436. #!ifdef WITH_NAT 437. # ----- rtpproxy params ----- 438. modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") 439. 440. # ----- nathelper params ----- 441. modparam("nathelper", "natping_interval", 30) 442. modparam("nathelper", "ping_nated_only", 1) 443. modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) 444. modparam("nathelper", "sipping_from", "sip:pinger@kamaz") 445. 446. # params needed for NAT traversal in other modules 447. modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") 448. modparam("usrloc", "nat_bflag", FLB_NATB) 449. #!endif 450. 451. 452. #!ifdef WITH_TLS 453. # ----- tls params ----- 454. modparam("tls", "config", "//etc/kamailio/tls.cfg") 455. #!endif 456. 457. #!ifdef WITH_ANTIFLOOD 458. # ----- pike params ----- 459. modparam("pike", "sampling_time_unit", 2) 460. modparam("pike", "reqs_density_per_unit", 16) 461. modparam("pike", "remove_latency", 4) 462. 463. # ----- htable params ----- 464. # ip ban htable with autoexpire after 5 minutes 465. modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") 466. #!endif 467. 468. #!ifdef WITH_XMLRPC 469. # ----- xmlrpc params ----- 470. modparam("xmlrpc", "route", "XMLRPC"); 471. modparam("xmlrpc", "url_match", "^/RPC") 472. #!endif 473. 474. #!ifdef WITH_DEBUG 475. # ----- debugger params ----- 476. modparam("debugger", "cfgtrace", 1) 477. #!endif 478. 479. ####### Routing Logic ######## 480. 481. 482. # Main SIP request routing logic 483. # - processing of any incoming SIP request starts with this route 484. # - note: this is the same as route { ... } 485. request_route { 486. 487. # per request initial checks 488. route(REQINIT); 489. 490. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); 491. route(DEBUG); 492. # NAT detection 493. route(NATDETECT); 494. 495. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in Route[NATDETECT] fix_nat-register\n"); 496. route(DEBUG); 497. 498. # CANCEL processing 499. if (is_method("CANCEL")) 500. { 501. if (t_check_trans()) { 502. route(RELAY); 503. } 504. exit; 505. } 506. 507. route(DEBUG); 508. 509. # handle requests within SIP dialogs 510. route(WITHINDLG); 511. 512. route(DEBUG); 513. 514. ### only initial requests (no To tag) 515. 516. t_check_trans(); 517. 518. route(DEBUG); 519. 520. # authentication 521. route(AUTH); 522. 523. route(DEBUG); 524. 525. # record routing for dialog forming requests (in case they are routed) 526. # - remove preloaded route headers 527. remove_hf("Route"); 528. if (is_method("INVITE|SUBSCRIBE")) 529. record_route(); 530. 531. # account only INVITEs 532. if (is_method("INVITE")) 533. { 534. setflag(FLT_ACC); # do accounting 535. } 536. 537. # dispatch requests to foreign domains 538. route(SIPOUT); 539. 540. ### requests for my local domains 541. 542. # handle presence related requests 543. route(PRESENCE); 544. 545. route(DEBUG); 546. 547. # handle registrations 548. route(REGISTRAR); 549. 550. route(DEBUG); 551. 552. if ($rU==$null) 553. { 554. # request with no Username in RURI 555. sl_send_reply("484","Address Incomplete"); 556. exit; 557. } 558. 559. # dispatch destinations to PSTN 560. # route(PSTN); 561. 562. route(DEBUG); 563. 564. # user location service 565. route(LOCATION); 566. 567. route(DEBUG); 568. 569. route(RELAY); 570. 571. } 572. 573. 574. route[RELAY] { 575. 576. # enable additional event routes for forwarded requests 577. # - serial forking, RTP relaying handling, a.s.o. 578. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { 579. if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); 580. } 581. if (is_method("INVITE|SUBSCRIBE|UPDATE")) { 582. if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); 583. } 584. if (is_method("INVITE")) { 585. if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); 586. } 587. 588. if (!t_relay()) { 589. sl_reply_error(); 590. } 591. exit; 592. } 593. 594. # Per SIP request initial checks 595. route[REQINIT] { 596. #!ifdef WITH_ANTIFLOOD 597. # flood dection from same IP and traffic ban for a while 598. # be sure you exclude checking trusted peers, such as pstn gateways 599. # - local host excluded (e.g., loop to self) 600. if(src_ip!=myself) 601. { 602. if($sht(ipban=>$si)!=$null) 603. { 604. # ip is already blocked 605. xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); 606. exit; 607. } 608. if (!pike_check_req()) 609. { 610. xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); 611. $sht(ipban=>$si) = 1; 612. exit; 613. } 614. } 615. #!endif 616. 617. if (!mf_process_maxfwd_header("10")) { 618. sl_send_reply("483","Too Many Hops"); 619. exit; 620. } 621. 622. if(!sanity_check("1511", "7")) 623. { 624. xlog("Malformed SIP message from $si:$sp\n"); 625. exit; 626. } 627. } 628. 629. # Handle requests within SIP dialogs 630. route[WITHINDLG] { 631. if (has_totag()) { 632. # sequential request withing a dialog should 633. # take the path determined by record-routing 634. if (loose_route()) { 635. route(DLGURI); 636. if (is_method("BYE")) { 637. setflag(FLT_ACC); # do accounting ... 638. setflag(FLT_ACCFAILED); # ... even if the transaction fails 639. } 640. else if ( is_method("ACK") ) { 641. # ACK is forwarded statelessy 642. route(NATMANAGE); 643. } 644. else if ( is_method("NOTIFY") ) { 645. # Add Record-Route for in-dialog NOTIFY as per RFC 6665. 646. record_route(); 647. } 648. route(RELAY); 649. } else { 650. if (is_method("SUBSCRIBE") && uri == myself) { 651. # in-dialog subscribe requests 652. route(PRESENCE); 653. exit; 654. } 655. if ( is_method("ACK") ) { 656. if ( t_check_trans() ) { 657. # no loose-route, but stateful ACK; 658. # must be an ACK after a 487 659. # or e.g. 404 from upstream server 660. route(RELAY); 661. exit; 662. } else { 663. # ACK without matching transaction ... ignore and discard 664. exit; 665. } 666. } 667. sl_send_reply("404","Not here"); 668. } 669. exit; 670. } 671. } 672. 673. # Handle SIP registrations 674. route[REGISTRAR] { 675. if (is_method("REGISTER")) 676. { 677. if(isflagset(FLT_NATS)) 678. { 679. setbflag(FLB_NATB); 680. # uncomment next line to do SIP NAT pinging 681. setbflag(FLB_NATSIPPING); 682. } 683. if (!save("location")) 684. sl_reply_error(); 685. 686. 687. #!ifdef WITH_ASTERISK 688. route(REGFWD); 689. #!endif 690. 691. exit; 692. } 693. } 694. 695. 696. #!ifdef WITH_ASTERISK 697. # Test if coming from Asterisk 698. route[FROMASTERISK] { 699. if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; 700. return -1; 701. } 702. 703. 704. # Send to Asterisk 705. route[TOASTERISK] { 706. $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); 707. xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); 708. route(RELAY); 709. exit; 710. } 711. 712. # Forward REGISTER to Asterisk 713. route[REGFWD] { 714. if(!is_method("REGISTER")) 715. { 716. return; 717. } 718. 719. xlog("[$mi] PRINT AU: $au PRINT fU: $fU Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 720. 721. t_relay_to_udp("50.0.0.10", "5060"); 722. 723. xlog("[$mi] PRINT AU: $au PRINT fU: $fU Sended SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 724. 725. } 726. #!endif 727. 728. # USER location service 729. route[LOCATION] { 730. 731. #!ifdef WITH_SPEEDDIAL 732. # search for short dialing - 2-digit extension 733. if($rU=~"^[0-9][0-9]$") 734. if(sd_lookup("speed_dial")) 735. route(SIPOUT); 736. #!endif 737. 738. #!ifdef WITH_ALIASDB 739. # search in DB-based aliases 740. if(alias_db_lookup("dbaliases")) 741. route(SIPOUT); 742. #!endif 743. 744. #!ifdef WITH_ASTERISK 745. if(is_method("INVITE") && (!route(FROMASTERISK))) { 746. # if new call from out there - send to Asterisk 747. # - non-INVITE request are routed directly by Kamailio 748. # - traffic from Asterisk is routed also directy by Kamailio 749. route(TOASTERISK); 750. exit; 751. } 752. #!endif 753. 754. $avp(oexten) = $rU; 755. if (!lookup("location")) { 756. $var(rc) = $rc; 757. route(TOVOICEMAIL); 758. t_newtran(); 759. switch ($var(rc)) { 760. case -1: 761. case -3: 762. send_reply("404", "Not Found"); 763. exit; 764. case -2: 765. send_reply("405", "Method Not Allowed"); 766. exit; 767. } 768. } 769. 770. # when routing via usrloc, log the missed calls also 771. if (is_method("INVITE")) 772. { 773. setflag(FLT_ACCMISSED); 774. } 775. 776. route(RELAY); 777. exit; 778. } 779. 780. # Presence server route 781. route[PRESENCE] { 782. if(!is_method("PUBLISH|SUBSCRIBE")) 783. return; 784. 785. if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { 786. route(TOVOICEMAIL); 787. # returns here if no voicemail server is configured 788. sl_send_reply("404", "No voicemail service"); 789. exit; 790. } 791. 792. #!ifdef WITH_PRESENCE 793. if (!t_newtran()) 794. { 795. sl_reply_error(); 796. exit; 797. } 798. 799. if(is_method("PUBLISH")) 800. { 801. handle_publish(); 802. t_release(); 803. } else if(is_method("SUBSCRIBE")) { 804. handle_subscribe(); 805. t_release(); 806. } 807. exit; 808. #!endif 809. 810. # if presence enabled, this part will not be executed 811. if (is_method("PUBLISH") || $rU==$null) 812. { 813. sl_send_reply("404", "Not here"); 814. exit; 815. } 816. return; 817. } 818. 819. # Authentication route 820. route[AUTH] { 821. 822. #!ifdef WITH_ASTERISK 823. # do not auth traffic from Asterisk - trusted! 824. if(route(FROMASTERISK)) 825. return; 826. #!endif 827. 828. 829. #!ifdef WITH_AUTH 830. 831. #!ifdef WITH_IPAUTH 832. if((!is_method("REGISTER")) && allow_source_address()) 833. { 834. # source IP allowed 835. return; 836. } 837. #!endif 838. 839. if (is_method("REGISTER") || from_uri==myself) 840. { 841. # authenticate requests 842. if (!auth_check("$fd", "subscriber", "1")) { 843. auth_challenge("$fd", "0"); 844. exit; 845. } 846. # user authenticated - remove auth header 847. if(!is_method("REGISTER|PUBLISH")) 848. consume_credentials(); 849. } 850. # if caller is not local subscriber, then check if it calls 851. # a local destination, otherwise deny, not an open relay here 852. if (from_uri!=myself && uri!=myself) 853. { 854. sl_send_reply("403","Not relaying"); 855. exit; 856. } 857. 858. #!endif 859. return; 860. } 861. 862. # Caller NAT detection route 863. route[NATDETECT] { 864. #!ifdef WITH_NAT 865. force_rport(); 866. if (nat_uac_test("19")) { 867. if (is_method("REGISTER")) { 868. fix_nated_register(); 869. } else { 870. if(is_first_hop()) 871. set_contact_alias(); 872. } 873. setflag(FLT_NATS); 874. 875. } 876. #!endif 877. return; 878. } 879. 880. # RTPProxy control 881. route[NATMANAGE] { 882. #!ifdef WITH_NAT 883. if (is_request()) { 884. if(has_totag()) { 885. if(check_route_param("nat=yes")) { 886. setbflag(FLB_NATB); 887. } 888. } 889. } 890. if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) 891. return; 892. 893. rtpproxy_manage(); 894. 895. if (is_request()) { 896. if (!has_totag()) { 897. if(t_is_branch_route()) { 898. add_rr_param(";nat=yes"); 899. } 900. } 901. } 902. if (is_reply()) { 903. if(isbflagset(FLB_NATB)) { 904. if(is_first_hop()) 905. set_contact_alias(); 906. } 907. 908. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[NATMANAGE] RTPproxy with EI Flags\n"); 909. 910. } 911. #!endif 912. return; 913. } 914. 915. # URI update for dialog requests 916. route[DLGURI] { 917. #!ifdef WITH_NAT 918. if(!isdsturiset()) { 919. handle_ruri_alias(); 920. 921. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[DLGURI] RTPproxy with EI Flags\n"); 922. 923. } 924. #!endif 925. return; 926. } 927. 928. # Routing to foreign domains 929. route[SIPOUT] { 930. if (!uri==myself) 931. { 932. append_hf("P-hint: outbound\r\n"); 933. route(RELAY); 934. } 935. } 936. 937. # PSTN GW routing 938. route[PSTN] { 939. #!ifdef WITH_PSTN 940. # check if PSTN GW IP is defined 941. if (strempty($sel(cfg_get.pstn.gw_ip))) { 942. xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); 943. return; 944. } 945. 946. # route to PSTN dialed numbers starting with '+' or '00' 947. # (international format) 948. # - update the condition to match your dialing rules for PSTN routing 949. if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) 950. return; 951. 952. # only local users allowed to call 953. if(from_uri!=myself) { 954. sl_send_reply("403", "Not Allowed"); 955. exit; 956. } 957. 958. if (strempty($sel(cfg_get.pstn.gw_port))) { 959. $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); 960. } else { 961. $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" 962. + $sel(cfg_get.pstn.gw_port); 963. } 964. 965. route(RELAY); 966. exit; 967. #!endif 968. 969. return; 970. } 971. 972. # XMLRPC routing 973. #!ifdef WITH_XMLRPC 974. route[XMLRPC] { 975. # allow XMLRPC from localhost 976. if ((method=="POST" || method=="GET") 977. && (src_ip==127.0.0.1)) { 978. # close connection only for xmlrpclib user agents (there is a bug in 979. # xmlrpclib: it waits for EOF before interpreting the response). 980. if ($hdr(User-Agent) =~ "xmlrpclib") 981. set_reply_close(); 982. set_reply_no_connect(); 983. dispatch_rpc(); 984. exit; 985. } 986. send_reply("403", "Forbidden"); 987. exit; 988. } 989. #!endif 990. 991. # route to voicemail server 992. route[TOVOICEMAIL] { 993. #!ifdef WITH_VOICEMAIL 994. if(!is_method("INVITE|SUBSCRIBE")) 995. return; 996. 997. # check if VoiceMail server IP is defined 998. if (strempty($sel(cfg_get.voicemail.srv_ip))) { 999. xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); 1000. return; 1001. } 1002. if(is_method("INVITE")) { 1003. if($avp(oexten)==$null) 1004. return; 1005. $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) 1006. + ":" + $sel(cfg_get.voicemail.srv_port); 1007. } else { 1008. if($rU==$null) 1009. return; 1010. $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) 1011. + ":" + $sel(cfg_get.voicemail.srv_port); 1012. } 1013. route(RELAY); 1014. exit; 1015. #!endif 1016. 1017. return; 1018. } 1019. 1020. # manage outgoing branches 1021. branch_route[MANAGE_BRANCH] { 1022. xdbg("new branch [$T_branch_idx] to $ru\n"); 1023. route(NATMANAGE); 1024. } 1025. 1026. # manage incoming replies 1027. onreply_route[MANAGE_REPLY] { 1028. xdbg("incoming reply\n"); 1029. if(status=~"[12][0-9][0-9]") 1030. route(NATMANAGE); 1031. } 1032. 1033. # manage failure routing cases 1034. failure_route[MANAGE_FAILURE] { 1035. route(NATMANAGE); 1036. 1037. if (t_is_canceled()) { 1038. exit; 1039. } 1040. 1041. #!ifdef WITH_BLOCK3XX 1042. # block call redirect based on 3xx replies. 1043. if (t_check_status("3[0-9][0-9]")) { 1044. t_reply("404","Not found"); 1045. exit; 1046. } 1047. #!endif 1048. 1049. #!ifdef WITH_VOICEMAIL 1050. # serial forking 1051. # - route to voicemail on busy or no answer (timeout) 1052. if (t_check_status("486|408")) { 1053. $du = $null; 1054. route(TOVOICEMAIL); 1055. exit; 1056. } 1057. #!endif 1058. } 1059. 1060. 1061. route[DEBUG] { 1062. if (method==("PUBLISH|SUBSCRIBE|REGISTER|OPTIONS")) { 1063. return; 1064. } 1065. xlog("[$mi] Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 1066. } -- С уважением, Александр Васин. 8 926 1437200 icq: 9906064

1 0

ngcp-mediaproxy-ng media address
by Kelvin Chua 03 Feb '14

03 Feb '14

somehow, mediaproxy is ignoring the media address set via rtpproxy_manage("co","10.17.0.102"); anybody tried this before? Jan 31 09:03:29 kam2 mediaproxy-ng[20455]: Got valid command from 127.0.0.1:46077: answer - { "sdp": "v=0#015#012o=root 345956998 345956998 IN IP4 10.17.0.105#015#012s=Asterisk PBX 11.7.0#015#012c=IN IP4 10.17.0.105#015#012t=0 0#015#012m=audio 15712 RTP/AVP 97 3 0 8 101#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=sendrecv#015#012", "replace": [ "session-connection", "origin" ], "call-id": "yqtupxcwsnbmcbo@X340saucy", "received-from": [ "IP4", "10.17.0.105" ], "media address": "22.9.18.15", "from-tag": "kjoel", "to-tag": "as478f57f3", "command": "answer" } Jan 31 09:03:29 kam2 mediaproxy-ng[20455]: Returning to SIP proxy: d3:sdp541:v=0#015#012o=root 345956998 345956998 IN IP4 10.17.0.102#015#012s=Asterisk PBX 11.7.0#015#012c=IN IP4 10.17.0.102#015#012t=0 0#015#012a=ice-lite#015#012m=audio 10002 RTP/AVP 97 3 0 8 101#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:10003#015#012a=ice-ufrag:QocOqhlR#015#012a=ice-pwd:HZ7WUgsHAKlLGp1n2FAzEwWsS9gj#015#012a=candidate:pMoXfVtH0dDfxpoa 1 UDP 2130706432 10.17.0.102 10002 typ host#015#012a=candidate:pMoXfVtH0dDfxpoa 2 UDP 2130706431 10.17.0.102 10003 typ host#015#0126:result2:oke Kelvin Chua

2 4

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Shankar 03 Feb '14

03 Feb '14

Hi Alex, We set the hash size to 32 MB and found that the hash size is not the same as configured one. At the max it was taking 64KB. Regards, Shankar Date: Mon, 03 Feb 2014 00:27:29 -0500 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regd. MAX_LDG_LOCKS Message-ID: <52EF28C1.5010204(a)evaristesys.com> Content-Type: text/plain; charset=windows-1252; format=flowed On 02/03/2014 12:22 AM, Shankar wrote: > Also we found that in ?dialog.c? the variable dlg_hash_size should be > ?unsigned int?. Hope this will be taken care in next release of Kamailio. How did you arrive at that conclusion? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, February 03, 2014 10:53 AM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Regd. MAX_LDG_LOCKS Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Also we found that in 'dialog.c' the variable dlg_hash_size should be 'unsigned int'. Hope this will be taken care in next release of Kamailio. Regards, Shankar

1 0

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Alex Balashov 03 Feb '14

03 Feb '14

On 02/03/2014 12:22 AM, Shankar wrote: > Also we found that in ‘dialog.c’ the variable dlg_hash_size should be > ‘unsigned int’. Hope this will be taken care in next release of Kamailio. How did you arrive at that conclusion? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

Regd. MAX_LDG_LOCKS
by Shankar 03 Feb '14

03 Feb '14

Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Also we found that in 'dialog.c' the variable dlg_hash_size should be 'unsigned int'. Hope this will be taken care in next release of Kamailio. Regards, Shankar

1 0

Kamailio behind DSL router
by Muhammad Shahzad 03 Feb '14

03 Feb '14

Hi, I am setting up Raspberry Pi to run Kamailio with mediaproxy-ng. This machine is running on local LAN behind a DSL router. Using dynamic DNS and DMZ services of router, i can access this box from the Internet. However, i do not know how to define advertised_address parameter to public IP of router. The public IP is dynamic and changes roughly every 12 hours or so. Is there any way kamailio can automatically learn this dynamic public IP and use that as advertised_address? Something similar to Asterisk's res_stun_monitor module, https://reviewboard.asterisk.org/r/854/ Any help will be greatly appreciated. Thank you.

2 6

Packet loss / Jitter / Latency issues
by Wingsravi R 02 Feb '14

02 Feb '14

Dear Daniel and Kamailio'ns Greetings, I am working on kamailio server (4.0.x),whicch is integreted with media-proxy server (2.5.2) and it is all working fine in the point of SIP functionality. And i have a set-up like this below, with this when the calls (audio/video) sessions started i am experiencing latency,jitter, echo's in audio calls and a choppy ,pixelled video : KAMAILIO | Android SIP client (IMSDroid) ---- Wi-Fi router ---- Switch (level 2) ---- Wi-Fi router ---- Android SIP client (IMSDroid) (Total set-up runs on Intranet infrastructure). When i call between two SIP clients, Calls are getting established successfully and RTP flow is also relaying using Media-proxy server. But the audio/video performnce is not good enough. I am experiencing all this Latency , jitter , Choppy pixelled video. How can i resolve all this issues? What could be the problem ? Ofcourse SIP proxy is not responsible all this problems (as i know ), But is there anything can be done on Media-proxy server ? Or anything Buffer settings can be done on Kamailio server side (or anything else) ? Even i changed Shared memory and Package memories to 512MB and 16MB respectively. Anybody can guess what may be the problems that causes these issues ? PS: When i tried 'ping' Clients IP addresses in Kamailio server PC, there is 20-30% packet loss also. Please help me in resolving these issues. Any help will greatly appreciate. Regards, Ravi.

4 3

Re: [SR-Users] dialog list query
by Charles Chance 01 Feb '14

01 Feb '14

John, What db_mode are you using for dialog module? Are the entries in the db if you check manually? Are you able to fetch other data using avp_db_query? And do you see any errors in the log? Regards, Charles On 1 Feb 2014 16:24, "John Murray" <john.murray(a)skyracktelecom.com> wrote: > Hello, > > > > I am managing calls with the dialog module in kamailio 4.0.4. > > > > However if is try to get the callid and from_tag of existing calls using: > > > > avp_db_query("select callid, from_tag from dialog", "$avp(s:s_callid), > $avp(s:s_from_tag)"); > > > > I get NULL, yet if I use: > > > > sercmd proxy dlg.list > > > > I get the calls listed correctly. > > > > What am I doing wrong? > > > > Thanks > > > > John > > _______________________________________________ > 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 > > -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

1 0

dialog list query
by John Murray 01 Feb '14

01 Feb '14

Hello, I am managing calls with the dialog module in kamailio 4.0.4. However if is try to get the callid and from_tag of existing calls using: avp_db_query("select callid, from_tag from dialog", "$avp(s:s_callid), $avp(s:s_from_tag)"); I get NULL, yet if I use: sercmd proxy dlg.list I get the calls listed correctly. What am I doing wrong? Thanks John

1 0

Kamailio without tls
by Neville D'Souza 01 Feb '14

01 Feb '14

Hi Support, I have recently kamailio 4.0.x version with tls on port 5061. Now our development team needs to connect to it using unencrypted on port 5060 to test an app for features and then work towards connecting through tls connection. So my question is how do I temporarily disable tls to test all the features on the kamailio server and then once feature testing is done revert back to tls connection? Best Regards, Neville D'Souza

2 1

Re: [SR-Users] [Kamailio-Business] Fosdem this year
by Ivo V 01 Feb '14

01 Feb '14

Hi, My 2 colleague and me (3 total) like to join the Kamailio dinner. We went to Beermania 2 years ago with the Jitsi guys and DanB and I can confirm the perfect fries! Ivo

4 6

kamailio 4.1.1 failure routes issue
by José Seabra 01 Feb '14

01 Feb '14

Hello there, I think that i found a bug within failure routes on version 4.1.1, i have a call flow that needs to use the failure routes 4 times if nobody answer the call, but when the call goes to the 3º attempt to same failure route kamailio doesn't handle the transaction, With the same kamailio script but using the version 4.0.1 kamailio works correctly. Please check the attachment with kamailio debug level 3 -- Cumprimentos José Seabra

1 0

Kamailio not mdifying Contact in multi-homed setup
by Biswas, Dipak 31 Jan '14

31 Jan '14

Hi All, I'm running into an issue, I'm not sure whether any of you seen this yourselves and resolved it. Please share some pointers. My network is: clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch) The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But, it's not. As a result, ACK from Caller is not reaching back to Kamailio. How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg. SIP 200 OK -> Via: SIP/2.0/UDP 10.1.128.11;branch=z9hG4bKa7ea.4013d9881c1b7fe7b4c6c0f0e8f9d6b6.0 Via: SIP/2.0/UDP <Caller Public IP>:5060;rport=5060;branch=z9hG4bK-383736-9b4621118533d3ccea17992738433249 Record-Route: <sip:10.1.128.11;r2=on;lr=on;ftag=791b5ae3;nat=yes> Record-Route: <sip:<Kamailio Pub IP>;r2=on;lr=on;ftag=791b5ae3;nat=yes> From: \"Dipak Biswas\" <sip:dipak.biswas@<Kamailio Pub IP>>;tag=791b5ae3 To: <sip:dipak.biswas.test2@<Kamailio Pub IP>>;tag=atXF5443gQj9p Call-ID: 20d5d6d366fcb06c259a0895b3e44b52@0:0:0:0:0:0:0:0 CSeq: 2 INVITE Contact: <sip:dipak.biswas.test2@10.1.128.34:5060;transport=udp> User-Agent: FreeSWITCH mod_sofia/1.4.2+git~20140108T200418Z~d8fc8469b4~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 269 -- Thanks, Dipak

1 0

sip reroute question
by Øystein Rudi 31 Jan '14

31 Jan '14

Hi, I have a scenario where multiple companies with multiple sip and smtp domains want to also use one common domain. Ie. Users from domain1.com have their own Exchange and Lync topology, this is also true for users from domain domain2.com Is there a way to use SER on commondomain.com to forward Lync traffic, presence, chat, rtp to these servers/users B.r, Øystein Rudi Teknisk leder (CTO) / Seljestadveien 78 A, 9407 Harstad sentralbord: +47 77 00 27 50 / fax: +47 77 00 27 51 / web: www.serit.no epost: oystein.rudi(a)itpartner.no / direkte: +47 77 00 27 81 / mobil: +47 93 00 89 99 [Serit_ITPartner_Harstad]

2 1

File transfering problem with RTPproxy
by Wingsravi R 31 Jan '14

31 Jan '14

Dear Daniel & Kamailio'ns I am working on File transferring feature between two SIP clients (IMSDroid). I have configured Kamailio (4.0.4) sever with RTPproxy to get the through NAT traversal of audio/videov calls. so fro that i added a script in kamailio.cfg as like below (Forced all INVITES through RTPproxy) : if (is_method("INVITE")) { setflag(7); # Set the qos flag #if (has_sdp()) { if(has_body("application/sdp")) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } #if (is_method("ACK") && has_sdp()) if (is_method("ACK") && has_body("application/sdp")) rtpproxy_answer(); .. .. onreply_route[1] { # if (has_sdp()) if(has_body("application/sdp")) rtpproxy_answer(); } onreply_route[2] { # if (has_sdp()) if(has_body("application/sdp")) rtpproxy_offer(); } But now when i am trying for File transferring (As per RFC 4975, Session mode IM) their INVITEs are also passing through RTPProxy service, which is dropping MSRP contents and with that Files are not getting transferring (But session is establishing between two clients). And when i tried file transferring without running RTPPproxy service. files are getting transferring. but that gives problems for audio/video calls. So anybody please help me in how to make it work for both MSRP&RTP packets. I mean how can i modify the script so that INVITE messges contains MSRP-RTP contents can be descriminate in the script, and then they can execute their respective blocks of scripts. And please find the attachments of my Kamailio.cfg file and file transfer traffic capture (tcpdump based) with RTPproxy instance running and without running. Any help will greatly appreciate. Awaiting for somebody's help. Regards Ravi

3 5

401 responses ending up in onreply_route instead of failure_route
by Daniel Tryba 31 Jan '14

31 Jan '14

Situation: endpoint -> kamailio (4.1.0) -> upstream sip server (kamailio 4.0.x) For an outbound call endpoint authenticates with kamailio. Kamailio relays the invite to upstream and upstream responds with a 401. This 401 should trigger authentication. So I thought I could use uac_auth() to fix this. uac_rely may only be used from FAILURE_ROUTE. So the initial INVITE from the endpoint has this set: if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } But to my suprise the 401 doesn't end up in the failure_route but in the onreply_route. Why? What am I missing? -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 4

Re: [SR-Users] Kamailio without tls
by Neville D'Souza 31 Jan '14

31 Jan '14

And also let me know wat can cause voice break up during conversation. Our Server has an public IP but our clients are behind nat. From: Neville D'Souza [mailto:ncd@gogopay.com] Sent: Friday, January 31, 2014 5:45 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio without tls Hi Support, I have recently kamailio 4.0.x version with tls on port 5061. Now our development team needs to connect to it using unencrypted on port 5060 to test an app for features and then work towards connecting through tls connection. So my question is how do I temporarily disable tls to test all the features on the kamailio server and then once feature testing is done revert back to tls connection? Best Regards, Neville D'Souza

1 0

one audio
by bkyeyune 31 Jan '14

31 Jan '14

Hello;iam running kamailio 4 with two interfaces one public and the other privateam using the default kamailio.cfg file on ubuntu*scenario*register and route calls on asterisk through the proxy private interface on the lan interfaceregister and route calls on asterisk through the public interface if you are on the internetbe able to call from either lan or internet to any one registered on the lan or internet*Achieved *can make register and make calls between endponts both on the internet two way audiocalls from the internet endpoint to lan registered end point one way audiocalls from lan endpoint to internet endpoint no audiocalls from lan to lan endpoints no audio*help*configuration of rtpproxy rule to allow audio flow to all endpointslan to lan lan to internet internet to lanusing the default configuration file route -n0.0.0.0 public 0.0.0.0 UG 1 0 0 eth00.0.0.0 private 0.0.0.0 UG 2 0 0 eth1public 0.0.0.0 pub sunet U 0 0 0 eth0private 0.0.0.0 pri subnet U 0 0 0 eth1/etc/default/rtpproxyCONTROL_SOCK="-F -s udp:*:7722"EXTRA_OPTS="-l public/private -d DBUG:LOG_LOCAL0 -u rtpproxy" -- View this message in context: http://sip-router.1086192.n5.nabble.com/one-audio-tp124687.html Sent from the Users mailing list archive at Nabble.com.

1 0

Notify with Websocket problem
by David Claybourne 31 Jan '14

31 Jan '14

Hi Kamailio-Community, i have the following problem. Publish and subscribe works as suspected. But after saving the presence state from a publish request and notifying the users via NOTIFY, users with a websocket connection don't get the notify request at all :(. Here is the version of kamailio i use: version: kamailio 4.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. I created a log file and attached the notify event (the problem is that a websocket dont have a real IP Address), is there any possibility to use the nathelper module for fixing this problem? I also attached the config file. Thanks for the help Greetings David Log: 9(25950) DEBUG: presence [notify.c:1470]: send_notify_request(): dialog info: 9(25950) DEBUG: presence [notify.c:117]: printf_subs(): pres_uri: sip:1001@example.com 9(25950) DEBUG: presence [notify.c:118]: printf_subs(): watcher_user@watcher_domain: 1000(a)example.com 9(25950) DEBUG: presence [notify.c:119]: printf_subs(): to_user@to_domain: 1001(a)example.com 9(25950) DEBUG: presence [notify.c:120]: printf_subs(): from_user@from_domain: 1000(a)example.com 9(25950) DEBUG: presence [notify.c:121]: printf_subs(): callid/from_tag/to_tag: sc19bmkpmluh91btkc4t7s(a)example.com/o0cqavak8u/6853f4be2e83d57dcdf85008ad4e4098-9f19 9(25950) DEBUG: presence [notify.c:122]: printf_subs(): local_cseq/remote_cseq: 2/4 9(25950) DEBUG: presence [notify.c:123]: printf_subs(): local_contact/contact: sip:5.45.101.128:8088;transport=ws/sip:1000@7b1g33n1ji.invalid 9(25950) DEBUG: presence [notify.c:124]: printf_subs(): record_route: 9(25950) DEBUG: presence [notify.c:125]: printf_subs(): sockinfo_str: tcp:5.45.101.128:8088 9(25950) DEBUG: presence [notify.c:126]: printf_subs(): event: presence 9(25950) DEBUG: presence [notify.c:127]: printf_subs(): status: active 9(25950) DEBUG: presence [notify.c:128]: printf_subs(): reason: 9(25950) DEBUG: presence [notify.c:129]: printf_subs(): version: 2 9(25950) DEBUG: presence [notify.c:130]: printf_subs(): expires: 595 9(25950) DEBUG: presence [notify.c:132]: printf_subs(): updated/updated_winfo: 0/0 9(25950) DEBUG: presence [notify.c:1563]: send_notify_request(): headers: Max-Forwards: 70 Event: presence Contact: <sip:5.45.101.128:8088;transport=ws> Subscription-State: active;expires=595 Content-Type: application/pidf+xml 9(25950) DEBUG: presence [notify.c:925]: ps_build_dlg_t(): CONTACT = sip:1000@7b1g33n1ji.invalid 9(25950) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [5.45.101.128] == [5.45.101.128] 9(25950) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 8088 (advertise 0) matches port 8088 9(25950) DEBUG: presence [notify.c:1573]: send_notify_request(): expires 595 status 1 9(25950) DEBUG: presence [notify.c:1726]: shm_dup_cbparam(): === 22/8/37 9(25950) DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:1000@7b1g33n1ji.invalid> 9(25950) DEBUG: <core> [dns_cache.c:569]: _dns_hash_find(): dns_hash_find((0), 33), h=0 9(25950) DEBUG: <core> [dns_cache.c:569]: _dns_hash_find(): dns_hash_find(7b1g33n1ji.invalid(18), 1), h=745 9(25950) ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "7b1g33n1ji.invalid" 9(25950) ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "7b1g33n1ji.invalid" 9(25950) ERROR: tm [uac.c:266]: t_uac_prepare(): t_uac: no socket found 9(25950) ERROR: presence [notify.c:1590]: send_notify_request(): in function tmb.t_request_within 9(25950) ERROR: presence [notify.c:1677]: notify(): sending Notify not successful 9(25950) ERROR: presence [notify.c:1275]: publ_notify(): Could not send notify for presence Config: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### #LOAD MODULES NEEDED #!define WITH_AUTH #!define WITH_PRESENCE #!define WITH_MYSQL #!define WITH_TLS #!define WITH_WEBSOCKETS #!define WITH_DEBUG ##!define WITH_MSRP #!substdef "!MY_IP_ADDR!5.45.101.128!g" #!substdef "!MY_DOMAIN!5.45.101.128!g" #!substdef "!MY_WS_PORT!8088!g" #!substdef "!MY_WSS_PORT!8089!g" #!substdef "!MY_MSRP_PORT!9000!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:databasepassword@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="example.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif #listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif #!ifdef WITH_MSRP listen=MY_MSRP_ADDR #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 #syn_branch=0 # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT # tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_MSRP loadmodule "msrp.so" loadmodule "htable.so" loadmodule "cfgutils.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 120) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif #!ifdef WITH_MSRP # ----- htable params ----- modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; exit; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #!ifdef WITH_MSRP event_route[msrp:frame-in] { msrp_reply_flags("1"); if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) { xlog("L_WARN", "MSRP request received on $Rp\n"); msrp_reply("403", "Action-not-allowed"); exit; } if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } if (!www_authenticate("MY_DOMAIN", "subscriber", "$msrp(method)")) { if (auth_get_www_authenticate("MY_DOMAIN", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } if ($hdr(Expires) != $null) { $var(expires) = (int) $hdr(Expires); if ($var(expires) < MSRP_MIN_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Min-Expires: MSRP_MIN_EXPIRES\r\n"); exit; } else if ($var(expires) > MSRP_MAX_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Max-Expires: MSRP_MAX_EXPIRES\r\n"); exit; } } else { $var(expires) = MSRP_MAX_EXPIRES; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); $shtex(msrp=>$var(sessid)) = $var(expires) + 5; # - Use-Path: the MSRP address for server + session id $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" + $var(sessid) + ";tcp\r\n" + "Expires: " + $var(expires) + "\r\n"; msrp_reply("200", "OK", "$var(hdrs)"); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if ($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "Session-does-not-exist"); exit; } else if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); } else { msrp_reply("501", "Request-method-not-understood"); } } #!endif

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t_relay_to_tls cannot be found
by jaflong jaflong 31 Jan '14

31 Jan '14

Can anyone suggest why t_relay_to_tls cannot be found 0(15284) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command t_relay_to_tls i am using kamailio 4.1.1

2 1

Kamailio is not starting when it connects against remote mysql server
by José Seabra 30 Jan '14

30 Jan '14

Hi there, I'm having a strange behavior when i start kamailio, it start processes but after few seconds all processes die, <core> [main.c:1630]: main_loop(): main_loop: Cannot fork Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7354]: DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql:// xxxx:xxxx@10.0.30.15/kamailio Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '10.0.30.15' (4) Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: uri_db [checks.c:249]: uridb_db_init(): unable to connect to the database Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [sr_module.c:896]: init_mod_child(): init_mod_child(): Error while initializing module uri_db (/usr/local/lib64/kamailio/modules/uri_db.so) Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [pt.c:350]: fork_process(): ERROR: fork_process(): init_child failed for process 5, pid 7335, "udp receiver child=4 sock=10.0.20.7:5060 ( 213.58.213.226:5060)" Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: : <core> [main.c:1630]: main_loop(): main_loop: Cannot fork anyone has idea why it is happening, it was working before and we didn't any changes -- Cumprimentos José Seabra

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Re: [SR-Users] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Klaus, I have been making some tests just to be sure the network is not the problem. While using a simpler config file for Kamailio with just the TLS and NAT which does not involve Asterisk in the scenario, calls work properly between softphones. OTOH, using the template provided in the KB referenced in my first post disabling the Asterisk define, calls are connected properly but no audio flows between phones. Now I am in the process of trying to locate where is the problem by comparing how both files handle the NAT support. Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/23/14 08:12 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 23.01.2014 10:29, John Smith wrote: > > Hello Klaus, > > > > I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy > > Just for clarity: > > force_send_socket is for near_end NAT traversal of the SIP signaling, > whereas manage_rtpproxy() is for the NAT traversal (near end and far > end) of the RTP stream. > > > It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > > If rtpproxy uses only a single listen-IP, then these flags are not > needed. Only if you operate rtpproxy in bridge mode, then you need these > flags. Bridge mode is necessary if you do not have IP routing between > the internal network and the "virtual external" network, or if you want > to bridge between IPv4 and IPv6. > > > Currently audio does not flow back to the softphones, it gets lost at Kamailio. > > Actually it should get lost at rtpproxy. > > Please post a SIP trace: ngrep -Wbyline -q -t -P "" port 5060 > > and post the setup (external + internal IP addresses) (you can send it > privately to me or mangle the IP addresses if they are sensitive) > > regards > Klaus > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio with mohqueue module
by mark＠brightvoip.co.uk 30 Jan '14

30 Jan '14

Hi, I am developing a system using Kamailio 4.1.1 as a front-end to a group of Asterisk servers. All of the usual functionality (registations, inbound, outbound, internal calls) are fine. We need to add call queueing to the system. I looked at the mohqueue module and read that we can use mohq_process() to send inbound calls to a suitably defined queue, based on the incoming URI. My question is what is the process for managing/automating the call to mohq_retrieve() to send calls to agents? Do I need to track agents with XMPP presence, etc? Thanks, Mark Hall

2 1

tcp_keepalive, tcp_keepidle and tcp_connection_lifetime
by Owais ul Haq 30 Jan '14

30 Jan '14

Hello, I am trying to understand kamailio.cfg parameters. Can some one please help me in understanding what these parameters do exactly when set. tcp_keepalive, tcp_keepidle and tcp_connection_lifetime. I have a general understanding of tcp. Want to know exact role of these parameters in kamailio.cfg. Thanking in advance. Cheers

2 1

htable failed auth banning
by Salvatore Nuzzo 30 Jan '14

30 Jan '14

hey guys. I was trying to add the htable routing rules here: http://preview.tinyurl.com/nomqsls but I need some help figuring our where they should go. See, I really like how this guy bans users for 15 minutes for failing to auth 3 times in a row. Where and how should I add this to my kam config? Any help would be greatly appreciated. Thanks

3 2

Re: [SR-Users] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Andrew, I did not know this function. I am using it to determine the direction of the flow. Thank you > ----- Original Message ----- > From: Andrew Pogrebennyk > Sent: 01/24/14 06:51 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 01/23/2014 05:12 PM, Klaus Darilion wrote: > >> It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > > > > If rtpproxy uses only a single listen-IP, then these flags are not > > needed. Only if you operate rtpproxy in bridge mode, then you need these > > flags. Bridge mode is necessary if you do not have IP routing between > > the internal network and the "virtual external" network, or if you want > > to bridge between IPv4 and IPv6. > > John, > This function can be used to check the direction of every message: > http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296 > You might also need to append the record-route parameters to remember > the flags you have passed to the manage_rtpproxy() initially. > Based on the direction of the request and initial flags you can > determine what flags to use when calling manage_rtpproxy() for a given > in-dialog requests and reply. > > Hope this helps. > Andrew > > _______________________________________________ > 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

Re: [SR-Users] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Fred, calls are being bridged across two interfaces: one natted from the outside with one public IP and one directly connected to the Asterisk. > ----- Original Message ----- > From: Fred Posner > Sent: 01/24/14 06:55 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 01/24/2014 09:51 AM, Andrew Pogrebennyk wrote: > > On 01/23/2014 05:12 PM, Klaus Darilion wrote: > >>> It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > >> > >> If rtpproxy uses only a single listen-IP, then these flags are not > >> needed. Only if you operate rtpproxy in bridge mode, then you need these > >> flags. Bridge mode is necessary if you do not have IP routing between > >> the internal network and the "virtual external" network, or if you want > >> to bridge between IPv4 and IPv6. > > > > John, > > This function can be used to check the direction of every message: > > http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296 > > You might also need to append the record-route parameters to remember > > the flags you have passed to the manage_rtpproxy() initially. > > Based on the direction of the request and initial flags you can > > determine what flags to use when calling manage_rtpproxy() for a given > > in-dialog requests and reply. > > > > Hope this helps. > > Andrew > > > > Are the calls being bridged across two interfaces or is the Kamailio > just natted? (or is it both?) > > Fred Posner > The Palner Group, Inc. > 503-914-0999 (direct) > 954-472-2896 (fax) > > _______________________________________________ > 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

Problem Re-writing RURI
by Geoffrey Mina 30 Jan '14

30 Jan '14

I am trying to manually rewrite the RURI for a certain scenario. The RURI is currently: sip:110219546496707@12.12.12.12;npdi;rate=0.000880 I want to re-write so it's just: sip:110219546496707@12.12.12.12 I have the following snippet which isn't working: xlog("L_INFO","sip:$(ruri{uri.user})@$(ruri{uri.host})"); $ru = "sip:$(ruri{uri.user})@$(ruri{uri.host})"; t_relay(); the log spits out exactly what I want, but writing the $ru variable results in: Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "$(ruri{uri.host})" Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "$(ruri{uri.host})" Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [t_fwd.c:1773]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [tm.c:1502]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [t_reply.c:1067]: run_failure_handlers(): ERROR: run_failure_handlers: Error in run_top_route Any hint would be greatly appreciated.

2 1

Notice: changes to apt repositories
by Daniel-Constantin Mierla 29 Jan '14

29 Jan '14

Hello, the current server hosting the apt repositories for debian and ubuntu is scheduled for replacement. It hasn't been updated with new packages in a while. Courtesy of Sipwise, a new build system is under development and should be available in the near future. Meanwhile, we plan to clone the existing packages to another server and have them available till the new system is up and running. As all these involve operations requiring DNS change and file transfer, the repositories might be unavailable from time to time. Many thanks to Jon Bonilla that took care of building the deb packages so far and Inaki Baz Castillo for hosting the afferent repositories. Note also that all Kamailio packages for latest version are also available via the official repositories from Debian unstable (sid) - thanks to Victor Seva for the work on it. Perhaps testing will get them soon and with the next stable version, should be no longer needed to use our repositories by users of Debian. We plan to keep ours as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] Contents of sr-users digest...
by Stuart Feeser 28 Jan '14

28 Jan '14

Regarding the sem_post problem, here is an old email from Tomasz Gebarowski which is interesting because it states that he had a sem_post issue with cdp.so which he solved by modifying the modules/cdp/Makefile. Now if we skip forward to right now, I notice that cdp.so COMPILE and LINKS for me with no errors and I also notice that Tomasz' fix is in place, but other ims modules that also use semaphore do NOT link properly for me in ubuntu and are exhibiting the same *undefined symbol: sem_post* error Tomasz had years ago in his ubuntu compiles. I think (hope) this may be more than coincidence that these things are related??? I have tried Tomasz' fix with an ubuntu compile but I still get the sem_post error. I still think he is on to something here and I am NOT modifying the Makefile correctly. Any suggestions on specifics to test this theory are greatly welcome. Here is Tomansz' email..... ---------------------------------------------------------------------------------------------------------------------- https://lists.berlios.de/pipermail/openimscore-users/2011-December/007219.h… [OpenIMSCore-Users] PROBLEM WITH ICSCF AND SCSCF STARTING Tomasz Gebarowski gebarowski at gmail.com Thu Dec 29 22:19:55 CET 2011 Previous message: [OpenIMSCore-Users] Fwd: How are you... Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hello, I've just googled some old post from October 2011, describing an issue with crashing I-CSCF and S-CSCF. The log message was something like: 0(2142) ERROR: load_module: could not open module /opt/OpenIMSCore/ser_ims/ modules/cdp/cdp.so>: /opt/OpenIMSCore/ser_ims/modules/cdp/cdp.so: undefined symbol: sem_post I managed to fix this issue on Ubuntu 11.10. It seems that UBUNTU, required linking cdp.so against pthread. Below you can find a simple patch: Index: modules/cdp/Makefile =================================================================== --- Makefile (revision 1175) +++ Makefile (working copy) @@ -20,7 +20,7 @@ endif DEFS += -I/usr/include/libxml2 -LIBS += -L$(LOCALBASE)/lib -lxml2 -lrt +LIBS += -L$(LOCALBASE)/lib -lxml2 -lrt -lpthread include ../../Makefile.modules =================================================================== Regards, Stuart Feeser

1 0

Kamailio with Asterisk
by Chirag Desai 28 Jan '14

28 Jan '14

Hi all, I'm wondering if someone can help me get a little unstuck. I followed the guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And I now have kamailio running, but I have two problems. Number 1 When I do kamctl add user password and make two users, they register fine. But any attempt to call between the users times out. Number 2 Anytime I have my kamailo.cfg file begin as #!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_ASTERISK I am unable to start kamctl. I get the following error. ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed If I run kamailio I get ERROR: bad config file (3 errors) The moment I remove the define lines at the start, kamctl starts fine, but I can't authenticate with asterisk. Any ideas? Thanks Chirag

3 2

ser-0.9.6 bug report
by David Binderman 28 Jan '14

28 Jan '14

Hello there, I just ran the static analyser "cppcheck" over the source code of ser-0.9.6 It said many things, including 1. [parser/parse_uri.c:566] -> [parser/parse_uri.c:568]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? Source code is case 'r': case 'R': b=p; state=PR2_R; default: state=URI_PARAM_P; Suggest add missing break. 2. > [parser/parse_uri.c:624] -> [parser/parse_uri.c:624]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? > [parser/parse_uri.c:628] -> [parser/parse_uri.c:628]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? > [parser/parse_uri.c:631] -> [parser/parse_uri.c:631]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? > [parser/parse_uri.c:636] -> [parser/parse_uri.c:636]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? Duplicates. Regards David Binderman

2 1

Replacing DR Module with Perl Script
by Nick Cameo 27 Jan '14

27 Jan '14

Hello Everyone, We are currently using the DR module for routing and failover, and would like to replace it with equivalent functionality using perl script. What we would like is to: * Order the gateways in a strict list given some constraints (Completed) * Build the RURI with the gateway IP, and prefix from the dr_gateways table (Similar to what do_routing does with the FROM tag and dr_gateways information) * Failover to the next gateway in strict order. I would really like to know what our options are, and the most graceful/scalable way of implementing this. PS An example would be highly appreciated! Kind Regards, Nick

2 2

undefined symbol: sem_post
by Stuart Feeser 27 Jan '14

27 Jan '14

My exact problem is described here,<http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html> but the thread never explains how to solve the problem, *"here" = * http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html *Description of the issue: * 1. Installing Kamailio 4.1 (latest download) on a *64 bit* Linux Mint version 16 Petra 2. The system compiles cleanly. I have included MAKE and Make Install info below 3. *ims_registrar_pcscf.so will not load* 4. * ims_qos.so will not load* 5. As an FYI, cdp.so successfully loads. I mention this because cdp.so uses semaphore but does not have the sem_post problem. I compared cdp.so with ims_registrar.pcscf.soMakefiles to see if there was something obvious, but no luck. 6. As an FYI, I get the exact same error from a debian apt-get install, so using the canned Debian solution won't work for me either. 7. The config file I used is the one included with the system which I completed with some info just for testing. I have attached it purely as FYI but I don't think that is the issue. *Parts of the syslog messages relevant for issue* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_qos.so>: /lib64/kamailio/modules/*ims_qos.so:* *undefined symbol: sem_post* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_registrar_pcscf.so>: /lib64/kamailio/modules/ *ims_registrar_pcscf.so*: *undefined symbol: sem_post* *Relevent section of MAKE* CC (gcc) [M ims_qos.so] cdpeventprocessor.o CC (gcc) [M ims_qos.so] mod.o CC (gcc) [M ims_qos.so] rx_aar.o CC (gcc) [M ims_qos.so] rx_asr.o CC (gcc) [M ims_qos.so] rx_authdata.o CC (gcc) [M ims_qos.so] rx_avp.o CC (gcc) [M ims_qos.so] rx_str.o CC (gcc) [M ims_qos.so] stats.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so CC (gcc) [M ims_registrar_pcscf.so] async_reginfo.o CC (gcc) [M ims_registrar_pcscf.so] reg_mod.o CC (gcc) [M ims_registrar_pcscf.so] save.o CC (gcc) [M ims_registrar_pcscf.so] service_routes.o CC (gcc) [M ims_registrar_pcscf.so] ul_callback.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so CC (gcc) [M ims_registrar_scscf.so] api.o CC (gcc) [M ims_registrar_scscf.so] common.o CC (gcc) [M ims_registrar_scscf.so] config.o CC (gcc) [M ims_registrar_scscf.so] cxdx_avp.o CC (gcc) [M ims_registrar_scscf.so] cxdx_callbacks.o CC (gcc) [M ims_registrar_scscf.so] cxdx_sar.o CC (gcc) [M ims_registrar_scscf.so] lookup.o CC (gcc) [M ims_registrar_scscf.so] path.o CC (gcc) [M ims_registrar_scscf.so] registrar_notify.o CC (gcc) [M ims_registrar_scscf.so] reg_mod.o CC (gcc) [M ims_registrar_scscf.so] regpv.o CC (gcc) [M ims_registrar_scscf.so] regtime.o CC (gcc) [M ims_registrar_scscf.so] reply.o CC (gcc) [M ims_registrar_scscf.so] rerrno.o CC (gcc) [M ims_registrar_scscf.so] save.o CC (gcc) [M ims_registrar_scscf.so] server_assignment.o CC (gcc) [M ims_registrar_scscf.so] sip_msg.o CC (gcc) [M ims_registrar_scscf.so] stats.o CC (gcc) [M ims_registrar_scscf.so] userdata_parser.o CC (gcc) [M ims_registrar_scscf.so] usrloc_cb.o ---------------------------------------------------------------------------------------------------- *Make Install* LD (gcc) [M ims_icscf.so] ims_icscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_icscf.so install -m 755 ims_icscf.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_qos.so install -m 755 ims_qos.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_registrar_pcscf.so install -m 755 ims_registrar_pcscf.so //lib64/kamailio/modules

1 0

undefined symbol: sem_post
by Stuart Feeser 27 Jan '14

27 Jan '14

My exact problem is described here,<http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html>but the thread never explains how to solve the problem, *"here" = * http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html *Description of the issue: * 1. Installing Kamailio 4.1 (latest download) on a *64 bit* Linux Mint version 16 Petra 2. The system compiles cleanly. I have included MAKE and Make Install info below 3. *ims_registrar_pcscf.so will not load* 4. * ims_qos.so will not load* 5. As an FYI, cdp.so successfully loads. I mention this because cdp.so uses semaphore but does not have the sem_post problem. I compared cdp.so with ims_registrar.pcscf.so Makefiles to see if there was something obvious, but no luck. 6. As an FYI, I get the exact same error from a debian apt-get install, so using the canned Debian solution won't work for me either. 7. The config file I used is the one included with the system which I completed with some info just for testing. I have attached it purely as FYI but I don't think that is the issue. *Parts of the syslog messages relevant for issue* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_qos.so>: /lib64/kamailio/modules/*ims_qos.so:* *undefined symbol: sem_post* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_registrar_pcscf.so>: /lib64/kamailio/modules/ *ims_registrar_pcscf.so*: *undefined symbol: sem_post* *Relevent section of MAKE* CC (gcc) [M ims_qos.so] cdpeventprocessor.o CC (gcc) [M ims_qos.so] mod.o CC (gcc) [M ims_qos.so] rx_aar.o CC (gcc) [M ims_qos.so] rx_asr.o CC (gcc) [M ims_qos.so] rx_authdata.o CC (gcc) [M ims_qos.so] rx_avp.o CC (gcc) [M ims_qos.so] rx_str.o CC (gcc) [M ims_qos.so] stats.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so CC (gcc) [M ims_registrar_pcscf.so] async_reginfo.o CC (gcc) [M ims_registrar_pcscf.so] reg_mod.o CC (gcc) [M ims_registrar_pcscf.so] save.o CC (gcc) [M ims_registrar_pcscf.so] service_routes.o CC (gcc) [M ims_registrar_pcscf.so] ul_callback.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so CC (gcc) [M ims_registrar_scscf.so] api.o CC (gcc) [M ims_registrar_scscf.so] common.o CC (gcc) [M ims_registrar_scscf.so] config.o CC (gcc) [M ims_registrar_scscf.so] cxdx_avp.o CC (gcc) [M ims_registrar_scscf.so] cxdx_callbacks.o CC (gcc) [M ims_registrar_scscf.so] cxdx_sar.o CC (gcc) [M ims_registrar_scscf.so] lookup.o CC (gcc) [M ims_registrar_scscf.so] path.o CC (gcc) [M ims_registrar_scscf.so] registrar_notify.o CC (gcc) [M ims_registrar_scscf.so] reg_mod.o CC (gcc) [M ims_registrar_scscf.so] regpv.o CC (gcc) [M ims_registrar_scscf.so] regtime.o CC (gcc) [M ims_registrar_scscf.so] reply.o CC (gcc) [M ims_registrar_scscf.so] rerrno.o CC (gcc) [M ims_registrar_scscf.so] save.o CC (gcc) [M ims_registrar_scscf.so] server_assignment.o CC (gcc) [M ims_registrar_scscf.so] sip_msg.o CC (gcc) [M ims_registrar_scscf.so] stats.o CC (gcc) [M ims_registrar_scscf.so] userdata_parser.o CC (gcc) [M ims_registrar_scscf.so] usrloc_cb.o ---------------------------------------------------------------------------------------------------- *Make Install* LD (gcc) [M ims_icscf.so] ims_icscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_icscf.so install -m 755 ims_icscf.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_qos.so install -m 755 ims_qos.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_registrar_pcscf.so install -m 755 ims_registrar_pcscf.so //lib64/kamailio/modules

1 0

alias_db DID regex?
by Pete Ashdown 26 Jan '14

26 Jan '14

Is there a way to use regex or Asterisk style numerical patterns in alias_db or any other db routing modules? Compressing a block of a hundred DID numbers to one entry would be really nice.

4 6

direct2rupayan@gmail.com has indicated you're a friend. Accept?
by direct2rupayan＠gmail.com 25 Jan '14

25 Jan '14

1 0

Static forward, reverse proxy
by Diego Woitasen 25 Jan '14

25 Jan '14

Hi, I need to configure Kamailio to act as reverse/frontend proxy, with static routing using rtpproxy. It will receive the SIP messages using UDP and must forward the traffic to the backends using TCP. The only IP address that the endpoints must see is the IP address of Kamailio. I've been looking at different examples on the web but I can't found something similar and I can't figure out how to start reading the official docs. If anyone has an example, doc or a hint it will be welcome. Regards, Diego -- Diego Woitasen Linux and Open Source solutions architect at www.vhgroup.net

1 0

Re: [SR-Users] Kamailio behind NAT
by John Smith 24 Jan '14

24 Jan '14

Hello Klaus, I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? Currently audio does not flow back to the softphones, it gets lost at Kamailio. Thank you for your help > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/23/14 12:26 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Am 21.01.2014 17:33, schrieb John Smith: > > The next test has been to comment out the rtpproxy_manage at NATMANAGE function and to put it both at route[RELAY] and onreply(route) following your post in this list from January 2013:http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html. > > > > Now the media flows from Phone1 to Kamailio, from Kamailio to Asterisk and back, but it gets stuck at Kamailio. I cannot see it flow towards the public IP of the Phone2. > > > > The force_send_socket you used could be of any use here? > That's what I recommend: > > - use 2 sockets, one for communication with internal nodes, one for > external clients > - in your Kamailio config check the direction of every message: i->e or > e->i (for requests and responses). Depending on the direction set the > proper IP when calling manage_rtpproxy and force the send socket: > > regards > Klaus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

4 3

pv_printf buffer overflow
by Aman Agarwal 24 Jan '14

24 Jan '14

I am running Kamailio-4.1.0 and I am getting this error If I try to parse JSON string longer than 1024 characters. ERROR: <core> [pvapi.c:1338]: pv_printf(): no more space for spec value ERROR: <core> [pvapi.c:1347]: pv_printf(): buffer overflow -- increase the buffer size... ERROR: <core> [sr_module.c:1696]: get_str_fparam(): Could not convert the PV-formated string to str ERROR: json [json_funcs.c:42]: json_get_field(): cannot get json string value I searched online for a fix and it was suggested to increase the PV_PRINT_BUF_SIZE in pvar.c. The source code for Kamailio 4.1.0 does not contain this file. I did not find this constant in other files too . Can some please suggest how to increase the buffer size? Thanks

2 1

Replicate registration
by Volkan Oransoy 23 Jan '14

23 Jan '14

Hi all, I am trying to setup an active-active pair of sip proxies. I have connected two kamailio boxes to the same PostgreSQL database and my usrloc db_mode is 3. I can see the registration data in the database but "ul show" outputs of two device is not same. I think writing registration data to the same db is not enough. I have found a method to replicate the registraiton process. With t_replicate, I synchronize REGISTER requests to both boxes. But I had to disable shared database because of uniqe id validations. route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > } > if (!save("location")) > sl_reply_error(); > t_replicate("sip:10.1.1.2:5060;transport=udp"); > exit; > } > } My media servers randomly sends calls to SIP proxies. If Media server send the call to the original registrar then the replicated, call initiates normally. But when UA registered in first proxy, and Media server sends the call to the send one, even if second one has UA in location table, call does not initiate. Does anyone have an idea about this kind of an active active setup? Thanks in advance. Best regards, /Volkan

4 4

P_usrloc - expired locations
by Helena Garcia-Nieto 23 Jan '14

23 Jan '14

Hi All, I am testing the p_usrloc module in order to write the location information of my kamailo server (Kamailio is on version 4.0.2) in two db As first test both DB are on the same machine, as well as the master db. On final version master and one db would probably be on the same machine and the second db on a different one. I have read on the p_usrloc module the following note: The p_usrloc module still has some missing feature, like automatic expiry of contacts and dumping of all users via the fifo cmd. And I can sadly see that the locations are not being deleted automatically after expiring. My users are mobile apps, that are programmed to be always logged in, and reconnecting to sip and IM servers if the devices detects a drop on the connection (bad wi-fi connection), or the user just move between different 3G cells. So the locations are most likely not going to be properly "unregistered" (expire 0) Have anyone found a solution or can give me some advice on how can I force the expired locations to be erased from the db, just as the usrloc module does? Thanks in advanced! Helena

2 5

Kamailio with extern Presence and XCAP server
by mohammed alyaseen 23 Jan '14

23 Jan '14

Hi all, Since 2 weeks I am trying to get Kamailio presence and xcap servers working. I am testing with RCS clients (native android clients). Somehow the presence server of kamailio doesn't like some subscribe events of RCS for example "Event: xcap-diff;path="resource-lists/users/sip:22222@kamailio.test/index", when RCS sends subscribe with this event, kamailio returns "Bad Event"... Another problem is some AUIDs used by RCS in the HTTP GETs for example ext-resource-lists. Kamailio doesn't support such as AUIDs and replies with not supported auid error.... Now I want to use extern presence and xdms servers, which run on the same virtual machine, where Kamailio is running. What should I do to achieve this? Any advice would be appreciated... Thanks Medo

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 23 Jan '14

23 Jan '14

Hi Shankar, I will setup and test case and try to reproduce as soon as I can. I suspect I will be able to get back to you early next week only Cheers Jason On Thu, Jan 23, 2014 at 8:48 AM, Shankar <shankar.rk(a)plintron.com> wrote: > Hi, > > > > From our repeated load tests what we can conclude irrespective of the > number of simultaneous calls, there is always this error “t_continue: > transaction not found” occurring. > > > > If I run say 20 cps, then after running 5000 calls, we observe exactly 20 > calls failing with the above error. We doubt that there is something > happening during a particular point in time (for a second) which impacts > the saving of those new transactions into shared memory. > > > > For 10 cps run, we observe exactly 10 call failures. We repeated with > different cps and found that error is exactly equal to the cps being run. > > > > Any configuration we are missing. Anyone can help? > > > > Regards, > > Shankar > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Tuesday, January 21, 2014 3:09 PM > *To:* 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express > Router (SER) - Users Mailing List' > *Subject:* FW: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Jason, > > > > Below is our config, > > > > route[LOCATION] { > > if(is_method("INVITE")) > > { > > if(!route(FROMCSCF)) > > { > > setflag(FLT_ACC); > > setflag(FLT_ACCFAILED); > > dlg_manage(); > > dlg_setflag("4"); > > > > async_route("RESUME", "1"); > > > > exit; > > } > > } > > } > > > > route[RESUME] > > { > > route(TO_LOCATION); // here t_relay to REGISTRAR is > done for user lookup. > > exit; > > } > > > > Regards, > > Shankar > > > > > > Date: Tue, 21 Jan 2014 11:14:21 +0200 > > From: Jason Penton <jason.penton(a)smilecoms.com> > > To: "Kamailio (SER) - Users Mailing List" > > <sr-users(a)lists.sip-router.org> > > Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > Message-ID: > > <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > We use it heavily, but not using the async module - we use it directly > from the IMS code. > > > > Can you please provide your config (or a relevant snippet) file so I can > see what exactly you are testing/trying to do > > > > Cheers > > jason > > > > *From:* Shankar [mailto:shankar.rk@plintron.com <shankar.rk(a)plintron.com>] > > *Sent:* Tuesday, January 21, 2014 2:25 PM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi, > > > > Anyone who had used t_suspend() and t_continue() can share the performance > details? > > > > I tried async module with one sec sleep time. I tried only 5 calls per > second but still it was not successful. After sometime I see below logs, > > > > Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: > t_continue(): ERROR: t_continue: transaction not found > > Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times > > Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times > > Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times > > > > Any configuration changes can help here? > > > > Regards, > > Shankar > > > > *From:* Shankar [mailto:shankar.rk@plintron.com <shankar.rk(a)plintron.com>] > > *Sent:* Wednesday, January 15, 2014 1:26 PM > *To:* 'Jason Penton' > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Jason, > > > > I am using 4.0.2 > > > > Regards, > > Shankar > > > > *From:* Jason Penton [mailto:jason.penton@smilecoms.com<jason.penton(a)smilecoms.com>] > > *Sent:* Wednesday, January 15, 2014 1:21 PM > *To:* Shankar > *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Subject:* Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Shankar, > > > > What version of Kamailio are you running? Kamailio -V > > > > Cheers > > Jason > > > > On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: > > Hi Jason, > > > > Please find below my response inline, > > > > > > I have some questions for you as we have used suspend/continue quite a lot > in the IMS code and don't have any leaks. > > > > Firstly, why are you using pkg_mem for your hash_id and label? Remember > that you will be in 2 different processes in the suspend and continue > portions of the code... so pkg_mem will not work - you should use shm_mem > instead. > > > > [Shankar] We use pkg_mem because we are invoking t_continue from the same > process ( using thread ). > > > > Secondly, how are you using top to tell that you have a leak? Kamailio's > memory is internally managed. > > > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the shared > usage of shared memory for the process. > > > > Cheers > > Jason > > > > > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > > > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com<shankar.rk(a)plintron.com> > ] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > > > > > > -- > > > > *Jason Penton* > > *Senior Manager: Applications and Services* > > *Smile Communications Pty (Ltd)* > > *Mobile:* > > +27 (0) 83 283 7000 > > *Skype:* > > jason.barry.penton > > jason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> > > www.smilecoms.com > > > > This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer > > > > > > > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

Re: [SR-Users] kamailio-4.0.2 core issue
by Daniel-Constantin Mierla 23 Jan '14

23 Jan '14

Hello, can you give the output in gdb for: bt full Also, do you have 'listen' parameters or kamailio is listening on all interfaces? It is recommended to upgrade to latest version 4.0.x, to be sure there is not a side effect of a fixed issues. Cheers, Daniel On 17/01/14 13:50, Surendra wrote: > Hi, > > We are facing a core issue in 4.0.2 version. Please find the GDB > trace below > > #0 udp_rcv_loop () at udp_server.c:471 > 471 fromlen=sockaddru_len(bind_address->su); > (gdb) bt > #0 udp_rcv_loop () at udp_server.c:471 > #1 0x0000000000475c14 in main_loop () at main.c:1851 > #2 0x000000000047a98b in main (argc=<value optimized out>, > argv=<value optimized out>) at main.c:2939 > > > We have simulated this so many times in load mode. > Machine details: Linux PLT-RA-RD-W167A.localdomain 2.6.18-308.el5 #1 > SMP Fri Jan 27 17:17:51 EST 2012 x86_64 x86_64 x86_64 GNU/Linux > > Can anyone help me on this > > > Regards > Surendra > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] SIP Update multi-homed
by Keith 23 Jan '14

23 Jan '14

Hi, Thanks for the replies so far, good news is I got it working by switching off mhomed (mhomed=0). However this seems to have broken my dispatcher as the SIP OPTIONS probing is being sent from the wrong interface! Any ideas how to set where dispatcher sends these from?? Cheers Keith

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Re: [SR-Users] Kamailio behind NAT
by John Smith 23 Jan '14

23 Jan '14

Hello Klaus, I have changed the rtpproxy_manage() in the route(NATMANAGE) to no avail. The c= in the Invite that arrives to Asterisk still points to the public IP of the Phone1. Code is as follows: route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (src_ip == ip.pbx.priva.te) { rtpproxy_manage(); } else { rtpproxy_manage("","ip.kamailio.priva.te"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } The next test has been to comment out the rtpproxy_manage at NATMANAGE function and to put it both at route[RELAY] and onreply(route) following your post in this list from January 2013: http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html. Now the media flows from Phone1 to Kamailio, from Kamailio to Asterisk and back, but it gets stuck at Kamailio. I cannot see it flow towards the public IP of the Phone2. The force_send_socket you used could be of any use here? Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/21/14 07:25 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Yes, when calling rtpproxy (whatever function you use, > manage/offer/answer/force), every function accepts as second parameter > the IP address which should be written into the SDP c=... line. > > So, detect the direction (eg based on source-ip) and put the respective > internal or external IP address into the SDP. > > Of course the NAT must be a static 1:1 NAT without any PAT. > > Btw: you have the same problem with some SIP headers, like Record-Route > or Via. Kamailio has to insert the respective public/private IP, > depending on where to send the message. If Kamailio uses only a single > socket, then you have to do this manually. The automatic approach is to > configure 2 listen=... sockets. One for the internal side, and one for > the external side. On the "external" socket you add the "advertise" > option - then Kamailio will automatically put the correct IP addresses > in RR and Via, eg: > > # public IP address of Kamailio/rtpproxy: 3.3.3.3 > # internal socket > listen=udp:10.10.0.2 > # external socket > listen=udp:10.10.0.3 advertise 3.3.3.3:5060 > mhomed=0 > > You could use separate subnets for the 2 sockets, or you could also use > only a single IP address with 2 different ports for the 2 sockets. > > > regards > Klaus > > > On 21.01.2014 14:50, John Smith wrote: > > The intended setup is as you described in option a. > > > > And the behaviour of rewriting SDP using private/public ports is what I understood the rtpproxy_manage was tasked to do. > > > > As it is not working the way I expected, what is the best way to proceed? To check if the IP is from the outside and then rewrite via rtpproxy_offer in the NATMANAGE block? > > > > Thank you > > > >> ----- Original Message ----- > >> From: Klaus Darilion > >> Sent: 01/21/14 05:25 AM > >> To: Kamailio (SER) - Users Mailing List > >> Subject: Re: [SR-Users] Kamailio behind NAT > >> > >> On 21.01.2014 13:24, John Smith wrote: > >>> I might be making wrong assumptions regarding this traffic flow. Is that correct? > >> > >> That depends on your policy. It is up to you to define how RTP should be > >> routed. There are basically 2 choices: > >> > >> a) RTP from clients is handled by rtpproxy: > >> > >> phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat-> phone2 > >> > >> In this case, only the private IP of Kamailio and rtpproxy (can be the > >> same IP address) must be mapped to a public IP address. > >> > >> > >> b) RTP directly to Asterisk: > >> > >> phone1 <-nat-> Asterisk <-nat-> phone2 > >> > >> In this case, the private IPs of Kamailio and Asterisk must be mapped to > >> a public IP address. > >> > >> > >> When using version a) you have to make sure to set the proper IP address > >> in the SDP. For example, SDPs in messages from Kamailio to the phone > >> must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in > >> messages from Kamailio to Asterisk must contain the PRIVATE IP of > >> rtpproxy in the c=... line. > >> > >> regards > >> Klaus > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 23 Jan '14

23 Jan '14

Hi, >From our repeated load tests what we can conclude irrespective of the number of simultaneous calls, there is always this error "t_continue: transaction not found" occurring. If I run say 20 cps, then after running 5000 calls, we observe exactly 20 calls failing with the above error. We doubt that there is something happening during a particular point in time (for a second) which impacts the saving of those new transactions into shared memory. For 10 cps run, we observe exactly 10 call failures. We repeated with different cps and found that error is exactly equal to the cps being run. Any configuration we are missing. Anyone can help? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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pdd value
by Oliver Roth 22 Jan '14

22 Jan '14

Hi all Question about an additional cdr information. We need to get the pdd value - means a time value till the user gets a ringback or an error- to track the delay, till a connection is done --> quality issue for customer. 5.6.1 Definition of Post Dial Delay Post Dial Delay (PDD) is experienced by the originating customer as the time from the sending of the final dialled digit to the point at which they hear ring tone or other in-band information. Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements. How can this be done in Kamailio? We use the carrierroute module with different fallback carriers. Kamailio version 3.3.5 Thanks in advance Oli

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Re: [SR-Users] SIP Update multi-homed
by Keith 21 Jan '14

21 Jan '14

Hi All, Thanks for all the advice, I managed to sort this one (I hope!). For anyone else interested I had to add some code at the start of the routing to look for a SIP UPDATE request. If it was a request then to set the $fs (forced socket) value to equal $Ri (Received interface) value. Once this was set it now holds the call open for longer than 30 seconds, fingers crossed it's all good! Thanks again, hope this is of some help to someone else. Keith

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Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

The intended setup is as you described in option a. And the behaviour of rewriting SDP using private/public ports is what I understood the rtpproxy_manage was tasked to do. As it is not working the way I expected, what is the best way to proceed? To check if the IP is from the outside and then rewrite via rtpproxy_offer in the NATMANAGE block? Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/21/14 05:25 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 21.01.2014 13:24, John Smith wrote: > > I might be making wrong assumptions regarding this traffic flow. Is that correct? > > That depends on your policy. It is up to you to define how RTP should be > routed. There are basically 2 choices: > > a) RTP from clients is handled by rtpproxy: > > phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat-> phone2 > > In this case, only the private IP of Kamailio and rtpproxy (can be the > same IP address) must be mapped to a public IP address. > > > b) RTP directly to Asterisk: > > phone1 <-nat-> Asterisk <-nat-> phone2 > > In this case, the private IPs of Kamailio and Asterisk must be mapped to > a public IP address. > > > When using version a) you have to make sure to set the proper IP address > in the SDP. For example, SDPs in messages from Kamailio to the phone > must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in > messages from Kamailio to Asterisk must contain the PRIVATE IP of > rtpproxy in the c=... line. > > regards > Klaus > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Using advertised IP address in manage_rttproxy would work with unpatched rtpproxy? Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/21/14 05:18 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 21.01.2014 12:27, Fred Posner wrote: > > With a patched version of rtpproxy you can advertise your private ip. > > > > http://www.fredposner.com/voip/1457/kamailio-behind-nat/ > > Aha, nice. Haven't known of this one. > > I always specified the "adverstised IP address" when calling > manage_rtpproxy(). That should work too. > > regards > Klaus > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Hello Kostas, the idea is to hide the Asterisk completely behind Kamailio, so I work only with the public IP of Kamailio and its SIP/RTP ports. There are no Asterisk RTP ports enabled, while the Kamailio RTP are already open. Thank you > ----- Original Message ----- > From: meres > Sent: 01/21/14 05:06 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Hi John, > > rtpproxy is not enough if you are using asterisk in your environment. > You have to check that asterisk is configured to work with NAT, otherwise you will experience audio problems. > Are the asterisk RTP ports enabled/forwarded on your firewall? > > Regards, > > Kostas > > On Jan 21, 2014, at 2:24 PM, John Smith <jsmith.15(a)mail.com> wrote: > > > Hi Fred, > > > > I have followed your HOWTO and the scenario remains exactly the same. > > > > I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk IP, and back directly to Phone2 public IP. > > > > I might be making wrong assumptions regarding this traffic flow. Is that correct? > > > > Thank you > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Hi Fred, I have followed your HOWTO and the scenario remains exactly the same. I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk IP, and back directly to Phone2 public IP. I might be making wrong assumptions regarding this traffic flow. Is that correct? Thank you

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Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Hello, I am currently deploying one Kamailio behind NAT with one Asterisk as explained in the Asipto KB (Kamailio 4.0.x and Asterisk 11.3.0 using Asterisk Database http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ). The structure is deployed as described in that document, with the only addition of one NAT between Kamailio and Internet: Phone ———————> Nat ———————> Kamailio ——————> Asterisk I have declared the private IP with the advertise option in order to support the NAT, enabled WITH_NAT and I have installed rtpproxy using standard Debian package configured as rtpproxy -l public_ip_ -s udp:localhost:7722 After setting up two phones which register correctly at Asterisk, I have no audio at all. By placing tcpdumps between nodes I see at Kamailio node both audio from public IP to internal Kamailio IP and from the latter to the Asterisk IP. In Asterisk I see audio coming from the Kamailio private IP and then back to the public IP of the phone. My guess is that audio should flow back into Kamailio and then to the phone, not directly from Asterisk as it is right now. Can anyone hint at where I am wrong? Thank you

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Re: [SR-Users] Regd. t_suspend() and t_continue()
by Shankar 21 Jan '14

21 Jan '14

Hi Daniel, Please find attached the log as requested. I ran 10 cps to reproduce it faster. Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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SIP Update multi-homed
by Keith 21 Jan '14

21 Jan '14

Hi, I have a kamailio server which is responding to SIP updates but sending from the wrong IP. I have multiple IPs in the same subnet on the same NIC. Is there anyway to say send update back out of the interface it was received on? Thanks. Keith

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Re: [SR-Users] SIP Update multi-homed
by Keith 21 Jan '14

21 Jan '14

Hi Daniel, So I have looked at the packet capture, the "SIP Request: UPDATE" comes in but when that request is forwarded to the phone it sends it from the interface IP and not the sub interface IP. Therefore the message never gets to the phone and the call is hung up after 30 seconds (due to the failure to respond). Cheers Keith

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FW: FW: Regd. t_suspend() and t_continue()
by Shankar 21 Jan '14

21 Jan '14

Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 21 Jan '14

21 Jan '14

We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason On Tue, Jan 21, 2014 at 10:54 AM, Shankar <shankar.rk(a)plintron.com> wrote: > Hi, > > > > Anyone who had used t_suspend() and t_continue() can share the performance > details? > > > > I tried async module with one sec sleep time. I tried only 5 calls per > second but still it was not successful. After sometime I see below logs, > > > > Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: > t_continue(): ERROR: t_continue: transaction not found > > Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times > > Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times > > Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times > > > > Any configuration changes can help here? > > > > Regards, > > Shankar > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Wednesday, January 15, 2014 1:26 PM > *To:* 'Jason Penton' > > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Jason, > > > > I am using 4.0.2 > > > > Regards, > > Shankar > > > > *From:* Jason Penton [mailto:jason.penton@smilecoms.com<jason.penton(a)smilecoms.com>] > > *Sent:* Wednesday, January 15, 2014 1:21 PM > *To:* Shankar > *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Subject:* Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Shankar, > > > > What version of Kamailio are you running? Kamailio -V > > > > Cheers > > Jason > > > > On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: > > Hi Jason, > > > > Please find below my response inline, > > > > > > I have some questions for you as we have used suspend/continue quite a lot > in the IMS code and don't have any leaks. > > > > Firstly, why are you using pkg_mem for your hash_id and label? Remember > that you will be in 2 different processes in the suspend and continue > portions of the code... so pkg_mem will not work - you should use shm_mem > instead. > > > > [Shankar] We use pkg_mem because we are invoking t_continue from the same > process ( using thread ). > > > > Secondly, how are you using top to tell that you have a leak? Kamailio's > memory is internally managed. > > > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the shared > usage of shared memory for the process. > > > > Cheers > > Jason > > > > > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > > > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com<shankar.rk(a)plintron.com> > ] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > > > > > > -- > > > > *Jason Penton* > > *Senior Manager: Applications and Services* > > *Smile Communications Pty (Ltd)* > > *Mobile:* > > +27 (0) 83 283 7000 > > *Skype:* > > jason.barry.penton > > jason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> > > www.smilecoms.com > > > > This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer > > > > > > > > > _______________________________________________ > 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 > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Daniel-Constantin Mierla 21 Jan '14

21 Jan '14

Hello, if you do: kamctl mi get_statistics all do you see an increase of active transactions counter? Would be possible for you to compile with MEMDBG=1 and then get the memory logs? make MEMDBG=1 cfg Then add your extra modules in include_modules variable inside modules.lst Afterwards: make install Edit kamailio.cfg and set memlog global parameter to a lower value than debug global parameter. Run kamailio for a while with the new config, when you see the shared memory keeps increasing, stop kamailio. Send the log messages from syslog to mailing list or made them available for download somewhere. Cheers, Daniel On 21/01/14 09:54, Shankar wrote: > > Hi, > > Anyone who had used t_suspend() and t_continue() can share the > performance details? > > I tried async module with one sec sleep time. I tried only 5 calls per > second but still it was not successful. After sometime I see below logs, > > Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm > [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found > > Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times > > Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times > > Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times > > Any configuration changes can help here? > > Regards, > > Shankar > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Wednesday, January 15, 2014 1:26 PM > *To:* 'Jason Penton' > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > Hi Jason, > > I am using 4.0.2 > > Regards, > > Shankar > > *From:*Jason Penton [mailto:jason.penton@smilecoms.com] > *Sent:* Wednesday, January 15, 2014 1:21 PM > *To:* Shankar > *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Subject:* Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > Hi Shankar, > > What version of Kamailio are you running? Kamailio -V > > Cheers > > Jason > > On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com > <mailto:shankar.rk@plintron.com>> wrote: > > Hi Jason, > > Please find below my response inline, > > I have some questions for you as we have used suspend/continue quite a > lot in the IMS code and don't have any leaks. > > Firstly, why are you using pkg_mem for your hash_id and label? > Remember that you will be in 2 different processes in the suspend and > continue portions of the code... so pkg_mem will not work - you should > use shm_mem instead. > > [Shankar] We use pkg_mem because we are invoking t_continue from the > same process ( using thread ). > > Secondly, how are you using top to tell that you have a leak? > Kamailio's memory is internally managed. > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the > shared usage of shared memory for the process. > > Cheers > > Jason > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com > <mailto:shankar.rk@plintron.com>> wrote: > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command > results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the > same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > > -- > > > *Jason Penton* > > *Senior Manager: Applications and Services* > > *Smile Communications Pty (Ltd)* > > *Mobile:* > > > > +27 (0) 83 283 7000 > > *Skype:* > > > > jason.barry.penton > > jason.penton(a)smilecoms.com <mailto:name.surname@smilecoms.com> > > www.smilecoms.com <http://www.smilecoms.com/> > > This email is subject to the disclaimer of Smile Communications athttp://www.smilecoms.com/disclaimer > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 21 Jan '14

21 Jan '14

Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: > Hi Jason, > > > > Please find below my response inline, > > > > > > I have some questions for you as we have used suspend/continue quite a lot > in the IMS code and don't have any leaks. > > > > Firstly, why are you using pkg_mem for your hash_id and label? Remember > that you will be in 2 different processes in the suspend and continue > portions of the code... so pkg_mem will not work - you should use shm_mem > instead. > > > > [Shankar] We use pkg_mem because we are invoking t_continue from the same > process ( using thread ). > > > > Secondly, how are you using top to tell that you have a leak? Kamailio's > memory is internally managed. > > > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the shared > usage of shared memory for the process. > > > > Cheers > > Jason > > > > > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > > > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com<shankar.rk(a)plintron.com> > ] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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Kamailio World 2014 - First group of speakers
by Daniel-Constantin Mierla 21 Jan '14

21 Jan '14

Hello, I would like to announce that the details about first group of speakers to present at Kamailio World Conference are now published at: - http://conference.kamailio.com/k02/speakers/ There are several excellent proposals still under review, we expect to fill the schedule very soon, hurry up if you intend to submit one. So far we have a good mixture of topics with a consistent group of new speakers comparing with the past edition. A selection with the titles of the proposals is available at: - http://conference.kamailio.com/k02/schedule/ Many thanks to the sponsors that make the event happening: FhG Fokus, Asipto, Sipwise, Sipgate, ITCenter, NG Voice. If you plan to attend, consider registering as soon as possible to secure your seat - registration form is available at: - http://conference.kamailio.com/k02/registration/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Non-authentication setup of Kamailio
by Carel Burger 20 Jan '14

20 Jan '14

Hi there, I have never used Kamailio before but want to investigate if if will work in my scenario before I invest time to learn it. I am the administrator of a small Wireless ISP. We do not provide SIP channels to our customers, we rather let them choose a service provider of their choice. Currently our customers are using two different providers. Since we are using ADSL with only one static IP we sometimes run into issues at the providers side with one way audio when our clients make a call to another client which is using the same service provider. I assume this is because of NAT. Since the RTP traffic actually leaves our network and then comes back to the other client the quality of the call is not as good as phoning a non client from the network. I want to know if it would be possible to setup Kamailio to keep the internal network calls traffic from leaving the network, and also allow free phone calls on our network. To do this we would require the authentication to take place on the service providers sip server and not on Kamailio. Kamailio would only do routing of the traffic and not worry about authentication. Would this be possible? Regards, Carel Burger

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Development of a Mobile VOIP app connecting to Kamailio Server
by Neville D'Souza 20 Jan '14

20 Jan '14

Hi, I recently got a Kamailio v4.0.0 server up and running with Siremis v4.0.0 on a Debian Wheezy system with mysql as backend database . The Chat, file sharing ,audio and video services work flawlessly even behind nat and the same need to be incorporated in an app. I have configured rtpproxy and tls with self signed certificates. This is basically a test environment and we will install the same kamailio version with commercial certificates on a live hosted server. Server side everything is working. Problem is on the development side. Our company is venturing into VOIP apps for the first time and trying to look at how to go about it. Firstly we would like to start with an android app and then move on to Iphone, Windows, Blackberry and Symbian app. We need to know what is required on server side (any software to be installed or any configuration on kamailio or addition of modules) and client side ( which library and development environment to use) to make it work. Lastly in terms of security what things I need to take care of in the server and app. Best Regards, Neville D'Souza

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kamailio crash after sqlops failures
by Uri Shacked 19 Jan '14

19 Jan '14

Daniel hi, I attached the following file: 1. "bt full" and "log" for each of the 3 servers. 2. The central MySQL slow queries log 3. The var/log/messages from one server. The servers crashed at different times but: 1. The MySQL server was very slow between 16:10:00 till 16:12:00 (approximately). 2. On each server, I am pretty sure, kamailio crashed while doing the data reload from modules like MTREE, HTABLE, DIALPLAN and CARRIERROUTE (all and all round 100,000 rows). Thanks again, Uri On Thu, Jan 16, 2014 at 2:17 AM, Daniel-Constantin Mierla <daniel(a)asipto.com > wrote: > Hello, > > the backtrace shows a crash in tm module, not sqlops. > > Can you tell which of the log files correspond to the instance that > produced the core file from where you took the back trace you attached? > > Can you give the backtraces from all cores? You say there were 3 crashes. > It is important to see if the backtrace is the same everywhere. > > Cheers, > Daniel >

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Re: [SR-Users] S-CSCF configuration brings to seg fault
by Gennaro Sarnataro 17 Jan '14

17 Jan '14

Hi Carsten, I would like to try my scenarios using a kamailio nightly build but i wasn't able to checkout it. Can you indicate me where to find latest builds? 2014/1/15 Gennaro Sarnataro <gen.sarnataro(a)gmail.com> > hi Carsten, > thank for your responsivness!!! > i'm using 4.1 version installed from repository. > how do you fix it?? > > > 2014/1/15 Carsten Bock <carsten(a)ng-voice.com> > >> Salut Gennaro, >> >> what Version of Kamailio are you using? Did you install it from GIT or >> from what repository? >> Looks similar to a bug, i just fixed yesterday... >> I recommend to use at least a 4.1 nightly build. >> >> Thanks, >> Carsten >> >> 2014/1/15 Gennaro Sarnataro <gsarnataro(a)it.tiscali.com>: >> > hello everyone, >> > I'm using some kamailio configurations to perform I-CSCF and S-CSCF. >> > My configurations file let me register and unregister correctly. >> > When i try to perform an invite I've a memory leak as you can see from >> this >> > log: >> > >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: >> received >> >> packet >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: >> received >> >> register >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth >> >> [authorize.c:663]: authenticate(): uri=sip:ims.test.net >> >> nonce=173692b1aea9182c2ac712f915f0144a >> >> response=f1efa37094e4492af1e292043c3d57ea qop=auth nc=00000001 >> >> cnonce=a2f5a827ac94467a39e81584c11e8284 hbody= >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth >> >> [authorize.c:708]: authenticate(): UE said: >> f1efa37094e4492af1e292043c3d57ea >> >> and we expect f1efa37094e4492af1e292043c3d57ea ha1 >> >> 1c31125e2929b3fd6a609bd1a134a631 (REGISTER) >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_registrar_scscf >> >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : >> >> [saa_return_code] - [-2] >> >> >> >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf >> >> [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: >> Failed >> >> finding avp >> >> >> >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf >> >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : >> >> [saa_return_code] - [1] >> >> >> >> Jan 15 11:07:50 localhost kamailio[2932]: WARNING: tm >> [t_lookup.c:1536]: >> >> t_unref(): WARNING: script writer didn't release transaction >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: ERROR: <script>: INVITE >> >> (sip:+393922568123@ims.test.net (10.139.140.249:14437) to >> >> sip:+39666555@ims.test.net, >> >> >> 0gQAAC8WAAACBAAALxYAAM3V1YJ1oc5VxmgXA6gfuMM+33iKMHHLO4FAjeWDRORBfoRu5acqCjeTKNa2pjdRdA--(a)10.139.140.249 >> ) >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: >> received >> >> packet >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: Mobile >> >> Originating Side >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Routing >> Mobile >> >> Originating side >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Adding angle >> >> brackets to P-Asserted-Identity >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: ims_isc [mod.c:236]: >> >> isc_match_filter(): Checking triggers >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: : <core> >> [mem/q_malloc.c:454]: >> >> qm_free(): BUG: qm_free: bad pointer 0x7f3dc03597a0 (out of memory >> block!) >> >> called from <core>: parser/parse_param.c: do_free_params(629) - >> aborting >> >> >> >> Jan 15 11:08:05 localhost kamailio[2943]: : <core> [pass_fd.c:293]: >> >> receive_fd(): ERROR: receive_fd: EOF on 90 >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:775]: >> >> handle_sigs(): child process 2940 exited by a signal 6 >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:778]: >> >> handle_sigs(): core was generated >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: <core> [main.c:790]: >> >> handle_sigs(): INFO: terminating due to SIGCHLD >> >> >> >> Jan 15 11:08:05 localhost kamailio[2939]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> >> >> ..... >> >> >> >> >> >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> Jan 15 11:08:05 localhost kamailio[2876]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:251]: >> >> cdp_exit(): CDiameterPeer child stopping ... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:428]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Terminating all >> >> children... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2939] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2938] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2937] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2936] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2935] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2934] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2933] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2932] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:447]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): All processes >> terminated. >> >> Cleaning up. >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:137]: >> >> worker_destroy(): Unlocking workers waiting on empty queue... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:140]: >> >> worker_destroy(): Unlocking workers waiting on full queue... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: CRITICAL: cdp >> >> [diameter_peer.c:487]: diameter_peer_destroy(): >> destroy_diameter_peer(): Bye >> >> Bye from C Diameter Peer test >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:253]: >> >> cdp_exit(): ... CDiameterPeer child stoped >> > >> > >> > after that the S-CSCF terminate. >> > it's unuseful to attach any wireshark dump because,as I told before, the >> > Serving process terminates just after receiving the INVITE. >> > Instead I attach my configuration. >> > >> > thanks in advance!!! >> > >> > saluti >> > >> > Gennaro Sarnataro >> > >> > _______________________________________________ >> > 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 >> > >> >> >> >> -- >> Carsten Bock >> CEO (Geschäftsführer) >> >> ng-voice GmbH >> Schomburgstr. 80 >> D-22767 Hamburg / Germany >> >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com >> >> Office +49 40 34927219 >> Fax +49 40 34927220 >> >> Sitz der Gesellschaft: Hamburg >> Registergericht: Amtsgericht Hamburg, HRB 120189 >> Geschäftsführer: Carsten Bock >> Ust-ID: DE279344284 >> >> Hier finden Sie unsere handelsrechtlichen Pflichtangaben: >> http://www.ng-voice.com/imprint/ >> >> _______________________________________________ >> 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 >> > > > > -- > Gennaro Sarnataro > -- Gennaro Sarnataro

2 1

Re: [SR-Users] S-CSCF configuration brings to seg fault
by Gennaro Sarnataro 17 Jan '14

17 Jan '14

hi Carsten, thank for your responsivness!!! how do you fix it?? i'm using 4.1 version installed from repository. I would like to try my scenarios using a kamailio nightly build but i wasn't able to checkout it. Can you indicate me where to find latest builds? 2014/1/15 Carsten Bock <carsten(a)ng-voice.com> > Salut Gennaro, > > what Version of Kamailio are you using? Did you install it from GIT or > from what repository? > Looks similar to a bug, i just fixed yesterday... > I recommend to use at least a 4.1 nightly build. > > Thanks, > Carsten > > 2014/1/15 Gennaro Sarnataro <gsarnataro(a)it.tiscali.com>: > > hello everyone, > > I'm using some kamailio configurations to perform I-CSCF and S-CSCF. > > My configurations file let me register and unregister correctly. > > When i try to perform an invite I've a memory leak as you can see from > this > > log: > > > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: > received > >> packet > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: > received > >> register > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth > >> [authorize.c:663]: authenticate(): uri=sip:ims.test.net > >> nonce=173692b1aea9182c2ac712f915f0144a > >> response=f1efa37094e4492af1e292043c3d57ea qop=auth nc=00000001 > >> cnonce=a2f5a827ac94467a39e81584c11e8284 hbody= > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth > >> [authorize.c:708]: authenticate(): UE said: > f1efa37094e4492af1e292043c3d57ea > >> and we expect f1efa37094e4492af1e292043c3d57ea ha1 > >> 1c31125e2929b3fd6a609bd1a134a631 (REGISTER) > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_registrar_scscf > >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : > >> [saa_return_code] - [-2] > >> > >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf > >> [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: > Failed > >> finding avp > >> > >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf > >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : > >> [saa_return_code] - [1] > >> > >> Jan 15 11:07:50 localhost kamailio[2932]: WARNING: tm [t_lookup.c:1536]: > >> t_unref(): WARNING: script writer didn't release transaction > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: ERROR: <script>: INVITE > >> (sip:+393922568123@ims.test.net (10.139.140.249:14437) to > >> sip:+39666555@ims.test.net, > >> > 0gQAAC8WAAACBAAALxYAAM3V1YJ1oc5VxmgXA6gfuMM+33iKMHHLO4FAjeWDRORBfoRu5acqCjeTKNa2pjdRdA--(a)10.139.140.249 > ) > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: > received > >> packet > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: Mobile > >> Originating Side > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Routing Mobile > >> Originating side > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Adding angle > >> brackets to P-Asserted-Identity > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: ims_isc [mod.c:236]: > >> isc_match_filter(): Checking triggers > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: : <core> [mem/q_malloc.c:454]: > >> qm_free(): BUG: qm_free: bad pointer 0x7f3dc03597a0 (out of memory > block!) > >> called from <core>: parser/parse_param.c: do_free_params(629) - aborting > >> > >> Jan 15 11:08:05 localhost kamailio[2943]: : <core> [pass_fd.c:293]: > >> receive_fd(): ERROR: receive_fd: EOF on 90 > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:775]: > >> handle_sigs(): child process 2940 exited by a signal 6 > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:778]: > >> handle_sigs(): core was generated > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: <core> [main.c:790]: > >> handle_sigs(): INFO: terminating due to SIGCHLD > >> > >> Jan 15 11:08:05 localhost kamailio[2939]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> > >> ..... > >> > >> > >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> Jan 15 11:08:05 localhost kamailio[2876]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:251]: > >> cdp_exit(): CDiameterPeer child stopping ... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:428]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Terminating all > >> children... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2939] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2938] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2937] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2936] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2935] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2934] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2933] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2932] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:447]: > >> diameter_peer_destroy(): destroy_diameter_peer(): All processes > terminated. > >> Cleaning up. > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:137]: > >> worker_destroy(): Unlocking workers waiting on empty queue... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:140]: > >> worker_destroy(): Unlocking workers waiting on full queue... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: CRITICAL: cdp > >> [diameter_peer.c:487]: diameter_peer_destroy(): > destroy_diameter_peer(): Bye > >> Bye from C Diameter Peer test > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:253]: > >> cdp_exit(): ... CDiameterPeer child stoped > > > > > > after that the S-CSCF terminate. > > it's unuseful to attach any wireshark dump because,as I told before, the > > Serving process terminates just after receiving the INVITE. > > Instead I attach my configuration. > > > > thanks in advance!!! > > > > saluti > > > > Gennaro Sarnataro > > > > _______________________________________________ > > 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 > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 34927219 > Fax +49 40 34927220 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > _______________________________________________ > 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

kamailio-4.0.2 core issue
by Surendra 17 Jan '14

17 Jan '14

Hi, We are facing a core issue in 4.0.2 version. Please find the GDB trace below #0 udp_rcv_loop () at udp_server.c:471 471 fromlen=sockaddru_len(bind_address->su); (gdb) bt #0 udp_rcv_loop () at udp_server.c:471 #1 0x0000000000475c14 in main_loop () at main.c:1851 #2 0x000000000047a98b in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2939 We have simulated this so many times in load mode. Machine details: Linux PLT-RA-RD-W167A.localdomain 2.6.18-308.el5 #1 SMP Fri Jan 27 17:17:51 EST 2012 x86_64 x86_64 x86_64 GNU/Linux Can anyone help me on this Regards Surendra

1 0

kamailio-4.1.1 avp issue within failure routes
by José Seabra 17 Jan '14

17 Jan '14

Hi there, I upgraded my kamailio server from version 4.0.1 to 4.1.1, then i noticed that all my "AVP's variavels" that are initialized on initial "INVITE", are all set as null when call goes to failure route, so kamailio is not keeping the variavel values for same call(dialog) When it happens i see the following warning "WARNING: tm [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release transaction" Thanks for your help -- Regards José Seabra

3 11

Memory Leaks with Kamailio using SEAS module
by ziad habchi 17 Jan '14

17 Jan '14

Dear Support, I am having an "out of memory" problem while using kamailio with SEAS module. After using Kamailio with the SEAS module for few hours with a high load , I am getting "out of memory" error that I believe is caused from a memory leak. I would like your help interpreting this error. The output of kamailio -v is : version: kamailio 4.0.4 (i386/linux) cabe58 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 14:23:27 Dec 2 2013 with gcc 4.1.2 kamctl ps : Process:: ID=0 PID=31191 Type=attendant Process:: ID=1 PID=31192 Type=udp receiver child=0 sock=193.100.200.18:5070 Process:: ID=2 PID=31193 Type=udp receiver child=1 sock=193.100.200.18:5070 Process:: ID=3 PID=31194 Type=udp receiver child=2 sock=193.100.200.18:5070 Process:: ID=4 PID=31195 Type=udp receiver child=3 sock=193.100.200.18:5070 Process:: ID=5 PID=31196 Type=slow timer Process:: ID=6 PID=31197 Type=timer Process:: ID=7 PID=31198 Type=ctl handler Process:: ID=8 PID=31199 Type=MI FIFO Process:: ID=9 PID=31200 Type=SEAS Process:: ID=10 PID=31201 Type=tcp receiver (generic) child=0 Process:: ID=11 PID=31202 Type=tcp receiver (generic) child=1 Process:: ID=12 PID=31203 Type=tcp receiver (generic) child=2 Process:: ID=13 PID=31204 Type=tcp receiver (generic) child=3 Process:: ID=14 PID=31205 Type=tcp main process kamcmd pkg.stats: { entry: 0 pid: 31191 rank: 0 used: 72480 free: 4091840 real_used: 102448 } { entry: 1 pid: 31192 rank: 1 used: 80624 free: 4083696 real_used: 110592 } { entry: 2 pid: 31193 rank: 2 used: 80624 free: 4083696 real_used: 110592 } { entry: 3 pid: 31194 rank: 3 used: 80624 free: 4083696 real_used: 110592 } { entry: 4 pid: 31195 rank: 4 used: 80624 free: 4083696 real_used: 110592 } { entry: 5 pid: 31196 rank: -1 used: 2276560 free: 4083744 real_used: 17770256 } { entry: 6 pid: 31197 rank: -1 used: 1787296 free: 4083744 real_used: 17280992 } { entry: 7 pid: 31198 rank: -2 used: 77760 free: 4086544 real_used: 107744 } { entry: 8 pid: 0 rank: 0 used: 89920 free: 4074352 real_used: 119936 } { entry: 9 pid: 0 rank: 0 used: 0 free: 0 real_used: 0 } { entry: 10 pid: 31201 rank: 5 used: 140240 free: 4023728 real_used: 170560 } { entry: 11 pid: 31202 rank: 6 used: 140240 free: 4023712 real_used: 170576 } { entry: 12 pid: 31203 rank: 7 used: 140240 free: 4023856 real_used: 170432 } { entry: 13 pid: 31204 rank: 8 used: 140240 free: 4023712 real_used: 170576 } { entry: 14 pid: 31205 rank: -4 used: 3877344 free: 4030176 real_used: 19269056 } As you can see above , the output of the SEAS module is not showing (pid 31200) Moreover , I can see that the real_used value in the kamcmd core.shmmem keep on increasing , and free decrease until I finally get an error "out of memory" When my SEAS client exit the output of kamcmd core.shmmem : { total: 33554432 free: 16273440 used: 1787296 real_used: 17280992 max_used: 33553520 fragments: 8098 } The traffic is a simple SIP MESSAGE from a seagull simulator, with a 200OK reply from my SEAS client. I appreciate your support, Thank you in advance, Ziad Habchi

3 10

Help required
by Gerhard De Villiers-Mohr 17 Jan '14

17 Jan '14

Good day Can you please forward me a manual of some sort or assist with setting up the kamailio serimis 4.0 to accept registration of IP PBX systems or soft phones. I am very new to this and need a step by step instructions. Also I will be getting a sip trunk to use for testing purposes before we go live and need help with setting this up and routing calls from users to the sip trunk for outbound calls. Please assist me , this is very new to me. If you prefer to do a remote login session I will gladly provide you with the teamviewer session details. Sincerely. Description: Description: Description: gerhard mohr email signature

1 0

Fosdem this year
by Daniel-Constantin Mierla 16 Jan '14

16 Jan '14

Hello, not much about real time communications this year at Fosdem, but being a convenient venue for me, the plan at this moment is to go . I saw some notes online from various people in the community that planned to attend as well, maybe we can synchronize and have an informal meeting around and/or a dinner. Who is planning to go? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

9 13

Kamailio multi domain issue..
by arun Jayaprakash 16 Jan '14

16 Jan '14

Hello, I am using Kamailio with a multidomain set up. It is working well but for one issue: I first populate the /etc/hosts file with the domain name and the IP address ( and of course the domain table ) but the new domain does not take effect till I restart the kamailio. Does anyone know how I can overcome this problem as it is not possible for me to keep restarting the server each time a new user registers. Thanks for the help. Arun

2 3

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