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37511 discussions

Multiple Htable Reload
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, I have multiple Htables, Is it possible to Reload all the Htables at one time via jsonrpc: As of Now I am using the below command to Reload one htable at a time. {"jsonrpc": "2.0","method": "htable.reload","params": ["b"],"id": 123} Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

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Map Technique(Key value pair) in kamailio
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, Is there any way to use the Mapping technique in kamailio. For Example: ========== Key = 91, Value = India Key = 92, Value = Srilanka How to store Key Value Pairs into kamailio and how to Retrieve? If I pass 91 it should Retrieve India, Is it possible? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

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Passive Open TCP sessions increases and kamailio runs out of memory
by Niklas Larsson 11 Sep '17

11 Sep '17

Hi, I have a kamailio 4.4.5 (x86_64/linux), running as a proxy in front of asterisk, handling udp/tcp/tls registration, using udp towards asterisk. Over time memory is consumed and the Passive Open TCP sessions is increasing until we run out of memory and a restart is needed. Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f9d55184000, 34120); Free fragment not found! Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f9d55184000, 34120) called from tls: tls_init.c: ser_malloc(275), module: tls; Free fragment not found! This image contains graphs that show the behaviour: https://imgur.com/68PvmyA Should it behave like this? Is there a knob to twist to change behaviour? tcp / tls stuff from kamailio.cfg: enable_tls=yes tcp_max_connections=8196 tls_max_connections=8196 tcp_no_connect=yes # Don't try to connect - drop if there is no session tcp_connection_lifetime=3605 modparam("tcpops", "closed_event", 1) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) in the registrar route I have: tcp_keepalive_enable("60", "5", "5"); /niklas

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kamailio asterisk queue
by Жан Базаров 11 Sep '17

11 Sep '17

hello! how i can forward presence peer state to asterisk queue? Thank's

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Q: purpose of t_newtran() in kamailio-basic.cfg route[LOCATION]?
by Anthony Alba 09 Sep '17

09 Sep '17

In kamailio-basic.cfg route[LOCATION] we have the following: --snip-- $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: --snip-- What is the purpose of t_newtran() here? Since we have not hit t_relay() yet (and don't have a transaction) is this to ensure the caller sees/acks the 404/405? Earlier in the mailing list https://lists.kamailio.org/pipermail/sr-users/2015-April/088062.html I read, although this is in reply to a different situation: "Why do you need to create new transactions yourself with t_newtran()? Nonexotic applications of Kamailio usually don't require this. ..." # User location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($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; } Thanks AAlba

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Re: [SR-Users] append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

After I add route(DISPATCH) to branch_route branch calls are going throught asterisk # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); xnotice("BRANCH: $rm $ru [$ci]"); route(DISPATCH); route(NATMANAGE); } but if I set $ru = "sip:34200@domain.org"; append_branch("sip:34201@domain.org"); first main ($ru) INVITE resolve to contact of ip phone INVITE sip:34200@172.16.3.7:5060 but branch doesnt' resolve INVITE sip:34201@domain.org Is there any way to resolve it? -- Aydar A. Kamalov

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append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

Hello! I have a simple route route[HUNT] { $ru = "sip:34200@172.16.3.7:5060"; append_branch("sip:34201@192.168.2.71"); } 172.16.3.7 and 192.168.2.71 ip phones 192.168.10.54 is kamailio 192.168.10.47 is asterisk If I called to this route kamailio will call to 34200 through dispatcher via asterisk. But branch going directly to the ip phone. Is it possible to call all branches through dispatcher and asterisks? U 192.168.10.54:5060 -> 192.168.10.47:5060 INVITE sip:34200@172.16.3.7:5060 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.0. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. U 192.168.10.54:5060 -> 192.168.2.71:5060 INVITE sip:34201@192.168.2.71 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.1. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. -- Aydar A. Kamalov

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need some help about acc module
by Vidja Prashant 08 Sep '17

08 Sep '17

Hello, I have some feature required changes in acc module. In acc module we getting cdrs in database only which have METHOD=INVITE METHOD=BYE but in my setup I also get METHOD=OPTIONS so I need this entry also in acc table in kamailio database. so which changes i need to do in kamailio acc module.... or other...... my cdrs log I paste below which have mehod=options "Sep 7 07:16:53 sip /usr/local/sbin/kamailio[29264]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: request accounted: timestamp=1504783013;method=OPTIONS;from_tag=eUC9UgGrD;to_tag=j90H6yy5pU6Xc;call_id=2xJd0KCt-P;code=200;reason=OK;calling=46566464645;called=you;egressIP= kamailio.org;src_ip=1.0.0.127;to=sip:+918866213320@sip.call.org ;tag=j90H6yy5pU6Xc;from=sip:918888888888@localhost;contact=;pai=;uri=sip: you(a)kamailio.org;onnet=0;rate=0.000333333" Thanks.. # #

2 1

crash after using async_route()
by Vitaliy Aleksandrov 07 Sep '17

07 Sep '17

Hello kamailio list, Recently found a problem in my configuration that uses async_route() functionality. It crashes after several calls when wait_timer fires. #0 0xb74a8556 in raise () from /lib/libc.so.6 #1 0xb74a9d78 in abort () from /lib/libc.so.6 #2 0x08293ae2 in qm_free (qmp=0xad65d000, p=0x3d64692d, file=0xb6216a16 "tm: h_table.c", func=0xb621663c <__FUNCTION__.18751> "free_cell_helper", line=187, mname=0xb621664d "tm") at core/mem/q_malloc.c:471 #3 0xb613f103 in free_cell_helper (dead_cell=0xae2cd210, silent=0, fname=0xb6239ea5 "timer.c", fline=655) at h_table.c:187 #4 0xb61e7758 in wait_handler (ti=557858937, wait_tl=0xae2cd258, data=0xae2cd210) at timer.c:655 #5 0x0826a2cc in timer_list_expire (t=557858937, h=0xad6b9668, slow_l=0xad6ba144, slow_mark=312) at core/timer.c:874 #6 0x08267cb1 in timer_handler () at core/timer.c:939 #7 0x0826a4d3 in timer_main () at core/timer.c:978 #8 0x08069575 in main_loop () at main.c:1721 #9 0x080707ca in main (argc=11, argv=0xbf85f044) at main.c:2723 When crash happens, kamailio prints the following message: Sep 4 16:15:38 [18938]: : <core> [core/mem/q_malloc.c:469]: qm_free(): BUG: qm_free: bad pointer 0x70707553 (out of memory block!) called from tm: h_table.c: free_cell_helper(187) - aborting Also had a few crashes in retransmission_handler(): #0 0xb750b556 in raise () from /lib/libc.so.6 #1 0xb750cd78 in abort () from /lib/libc.so.6 #2 0xb6249b5a in retransmission_handler (r_buf=0xae036674) at timer.c:367 #3 0xb6247558 in retr_buf_handler (ticks=1234464444, tl=0xae036688, p=0x1f40) at timer.c:594 #4 0x0826a2cc in timer_list_expire (t=1234464444, h=0xad71c668, slow_l=0xad71cd44, slow_mark=2232) at core/timer.c:874 #5 0x08267cb1 in timer_handler () at core/timer.c:939 #6 0x0826a4d3 in timer_main () at core/timer.c:978 #7 0x08069575 in main_loop () at main.c:1721 #8 0x080707ca in main (argc=11, argv=0xbff64134) at main.c:2723 ERROR: tm [timer.c:366]: retransmission_handler(): transaction 0xae0365e0 scheduled for deletion and called from RETR timer (flags 6d) Both timers fired for an INVITE transaction that was previously suspended by async_route(), then resumed, sent out and received a 4xx reply (407). This configuration worked fine with kamailio 4.2.x and problem appeared after upgrading to 5.0.2. Trying to figure out how to narrow down the problem. Any input is appreciated.

2 6

alias_db muliple usernames for alias
by Aidar Kamalov 07 Sep '17

07 Sep '17

Hello, I have modparam("alias_db", "append_branches", 1) and table dbaliases +----+----------------+-----------------+----------+-----------------+ | id | alias_username | alias_domain | username | domain | +----+----------------+-----------------+----------+-----------------+ | 1 | 1111 | sipchel | 34200 | sipchel | | 2 | 1111 | sipchel | 34201 | sipchel | +----+----------------+-----------------+----------+-----------------+ If I call to 1111 ringing only 34200, because first invite is right (resolving to username contact), but branch inivite is wrong. Is it bug or it is my my fault? First: INVITE sip:34200@172.16.3.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.0 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> Second: INVITE sip:34201@sipchel SIP/2.0 Record-Route: <sip:192.168.10.54;lr=on;ftag=1898fba2a5729040o0;did=958.fd11> Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.1 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> -- Aydar A. Kamalov

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