sr-users December 2010

sr-users@lists.kamailio.org

63 participants
115 discussions

Recursive failure_route after CFNA
by Igor Potjevlesch 21 Dec '10

21 Dec '10

Hello all, I have setup a mechanism to forward outgoing calls to another gateway if the first one replies with 5xx. I have simply used ‘t_check_status’ in the failure_route. This works. But, with a CFNA it doesn’t work : the incoming call is treated by SER (0.9.9). After the timer has expired the call goes to ‘failure_route’ for the first time to setup the CFNA with, finally, an ‘append_branch’. At this moment, the first gateway replies with 5xx and the call never goes to ‘failure_route’. I can match the 5xx within the ‘onreply_route’, but that’s all. So, instead of trying another gateway, the initial incoming calls terminate on the voicemail. Can anyone help me? Regards, Igor.

2 2

SIP UA's behind same NAT
by Deon Vermeulen 20 Dec '10

20 Dec '10

Hi List Can someone please help me? I need to implement PBX Services for a customer that has 10 IP Phones at his Main site and then 20 IP Phones located over 5 Branches. My goal is to have RTP go end to end and not proxied. I had a look at the below link and tried it, but with no luck. http://kamailio.org/dokuwiki/doku.php/examples:caller-callee-behind-same-nat Here is a partial output of my kamailio.cfg file if someone would be so kind to help me. #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:localhost: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@x.x.x.x") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } Thank you in advance Kind Regards Deon

2 5

CRASH on qm_free for simultaneous calls (V3.1.0)
by Jijo 20 Dec '10

20 Dec '10

I'm observing a crash on qm_free, when we made two outgoing calls to the same number. One SIP phone receives ring back tone and the other busy tone. The back trace is shown below which shows that it is crashed in abort(). How can the condition if (f->u.is_free){ could be true? Please let me know if anybody observed this issue and let me know how to debug this isssue. f=(struct qm_frag*) ((char*)p-sizeof(struct qm_frag)); #ifdef DBG_QM_MALLOC qm_debug_frag(qm, f); if (f->u.is_free){ LOG(L_CRIT, "BUG: qm_free: freeing already freed pointer," " first free: %s: %s(%ld) - aborting\n", f->file, f->func, f->line);* abort();* } MDBG("qm_free: freeing frag. %p alloc'ed from %s: %s(%ld)\n", f, f->file, f->func, f->line); #endif BACKTRACE ------------------- Program received signal SIGABRT, Aborted. 0xb76fd7a5 in raise () from /lib/libc.so.6 (gdb) bt full #0 0xb76fd7a5 in raise () from /lib/libc.so.6 No symbol table info available. #1 0xb76ff070 in abort () from /lib/libc.so.6 No symbol table info available. #2 0x081824d5 in qm_free (qm=0xaf7ee000, p=0xaf9ed758, file=0xb7659aed "tm: h_table.c", func=0xb7659c9c "free_cell", line=141) at mem/q_malloc.c:447 f = <value optimized out> prev = <value optimized out> next = <value optimized out> size = <value optimized out> #3 0xb75ff6b9 in free_cell (dead_cell=0xaf9e7054) at h_table.c:141 b = <value optimized out> i = <value optimized out> rpl = <value optimized out> tt = <value optimized out> foo = <value optimized out> cbs = <value optimized out> cbs_tmp = <value optimized out> __FUNCTION__ = "free_cell" #4 0xb7625eb0 in t_unref (p_msg=0x847bb10) at t_lookup.c:1553 kr = <value optimized out> #5 0xb764b3ad in w_t_unref (foo=0x847bb10, flags=2147483649, bar=0x0) at tm.c:707 No locals. #6 0x0811543a in exec_post_script_cb (msg=0x847bb10, type=REQUEST_CB_TYPE) at script_cb.c:195 cb = 0x8867874 flags = 2147483649 #7 0x080e5a4e in receive_msg ( buf=0x82a89e0 "ACK sip:0845@10.80.13.54:5060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 10.200.3.39:5060;branch=z9hG4bK0a992e8d3052e85e5\r\nRoute: <sip:0845@10.200.0.31:5060;lr;transport=udp>\r\nMax-Forwards: 69\r\nFrom: 554"..., len=<value optimized out>, rcv_info=0xbf95b7fc) at receive.c:221 msg = 0x847bb10 ctx = {rec_lev = 137434160, run_flags = 0, last_retcode = -1080707352, jmp_env = {{__jmpbuf = {-1224488143, 137089696, 137434092, 0, -1080707248, -1080707304}, __mask_was_saved = 0, __saved_mask = {__val = {3079133424, 134556827, 3079002384, 0, 3214260024, 3070479287, 137089696, 137434160, 3070572886, 3070573396, 228, 137347864, 3214260056, 3077419768, 3079002384, 137089664, 4294967295, 3079131124, 134556827, 134535424, 1, 3079058382, 3079133864, 3079003192, 1, 1, 0, 134555968, 0, 136548964, 3077419768, 142823652}}}}} ret = <value optimized out> inb = { s = 0x82a89e0 "ACK sip:0845@10.80.13.54:5060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 10.200.3.39:5060;branch=z9hG4bK0a992e8d3052e85e5\r\nRoute: <sip:0845@10.200.0.31:5060;lr;transport=udp>\r\nMax-Forwards: 69\r\nFrom: 554"..., len = 457} __FUNCTION__ = "receive_msg" #8 0x08175cb6 in udp_rcv_loop () at udp_server.c:532 len = <value optimized out> buf = "ACK sip:0845@10.80.13.54:5060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 10.200.3.39:5060;branch=z9hG4bK0a992e8d3052e85e5\r\nRoute: <sip:0845@10.200.0.31:5060;lr;transport=udp>\r\nMax-Forwards: 69\r\nFrom: 554"... from = 0x88350e4 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {654559242, 0, 3214260312, 3079002384}, addr32 = {654559242, 0, 3214260312, 3079002384}, addr16 = { 51210, 9987, 0, 0, 47192, 49045, 55568, 46981}, addr = "\n\310\003'\000\000\000\000X\270\225\277\020Ù \267"}}, dst_ip = {af = 2, len = 4, u = {addrl = {520144906, 0, 0, 0}, addr32 = {520144906, 0, 0, 0}, addr16 = {51210, 7936, 0, 0, 0, 0, 0, 0}, addr = "\n\310\000\037", '\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\n\310\003'\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = { s_addr = 654559242}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 654559242, 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 = 0x82f4140, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #9 0x080ac1d4 in main_loop () at main.c:1554 i = <value optimized out> ---Type <return> to continue, or q <return> to quit--- pid = <value optimized out> si = <value optimized out> si_desc = "udp receiver child=0 sock=10.200.0.31:5060\000\b A\206\b\002\000\000\000\004\000\000\000\n\310\000\037L \225\257H\271\225\277@ \030\207\267\020i\203\b\006\000\000\000\000\340~\257\001\000\000\000\001\000\000\000\000\000\000\000\000\340~\257\004\000\000\000?)\037\b\001\000\000\000\a\000\000\000\000\000\000\000H\271\225\277\350\353\017\b" #10 0x080af2ec in main (argc=9, argv=0xbf95ba84) at main.c:2398 cfg_stream = 0x8cbe008 c = <value optimized out> r = <value optimized out> tmp = 0xbf95cdc0 "" tmp_len = -1217394389 port = <value optimized out> proto = <value optimized out> options = 0x81ef340 ":f:cm:dVhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 3337198521 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = 0 p = <value optimized out> Thanks Jijo

2 5

new: pkg monitoring and troubleshooting enhancements
by Daniel-Constantin Mierla 20 Dec '10

20 Dec '10

Hello, I have committed some code to allow easier troubleshooting and monitoring of PKG usage per process. One new functions are exported by cfgutils module: - pkg_summary() - can be used to dump summary of PKG usage Btw, there was also added shm_summary() that dumps the summary of SHM usage. The kex module exports a new rpc command 'pkg.stats' that prints PKG usage metrics (used, free, real_used) per process. Here is a sample output: sercmd> pkg.stats { entry: 0 pid: 4793 rank: 0 used: 124788 free: 3813512 real_used: 380792 } { entry: 1 pid: 4794 rank: 1 used: 132740 free: 3805512 real_used: 388792 } { entry: 2 pid: 4795 rank: 2 used: 132740 free: 3805512 real_used: 388792 } The adding free to real_used values results in pkg memory size per process (4MB in this example). To get these stats, you have to compile with MALLOC_STATS enabled (this is default for Kamailio, so nothing actually to do for it). One benefit of this pkg.stats is ability to monitor usage of private memory and trigger alerts if the amount of available memory (free) for one process goes below a lower limit, useful when there is a memory leak (so you restart in time and report the problem) or the config operations for given traffic needs larger PKG size. In the near future there will be some new additions to be able to access these values also via PVs. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

SER
by douglas ugalde 19 Dec '10

19 Dec '10

Can you tell me please what can I do to change the port on SER because I installed on Asterisk and SER listen on port 5060 right? so it blocked Asterisk when is running, please tell me how can I fix this problem. Regards.

2 1

Kamailio Compatability with Elastix
by Kurt Mullen 17 Dec '10

17 Dec '10

Well I finally made some progress. I have Kamailio 3.1.0 installed on one server and Elastix 2.0.3 installed on another server. The Kamailio server queries Elastix for authentication, and the SIP clients are able to register with Kamailio. However, the clients cannot call each other. Also, extensions added in the Elastix interface cannot authenticate with Kamailio. I have to manually add the users in the Asterisk database using mysql commands to get the authentication to work. I will need to use a GUI for staff to add / delete numbers. Am I way off base here? Kurt Mullen

1 0

/mysql/#-create.sql: No such file or directory
by Kurt Mullen 17 Dec '10

17 Dec '10

When attempting to run : /usr/local/sbin/kamdbctl create I get the following results: /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 155: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed! Contents of directory: root@Kamailio:/usr/local/share/kamailio/mysql# ls acc-create.sql drouting-create.sql purple-create.sql alias_db-create.sql group-create.sql registrar-create.sql auth_db-create.sql htable-create.sql rls-create.sql avpops-create.sql imc-create.sql siptrace-create.sql carrierroute-create.sql lcr-create.sql speeddial-create.sql cpl-create.sql matrix-create.sql standard-create.sql dialog-create.sql msilo-create.sql uac-create.sql dialplan-create.sql mtree-create.sql uri_db-create.sql dispatcher-create.sql pdt-create.sql userblacklist-create.sql domain-create.sql permissions-create.sql usrloc-create.sql domainpolicy-create.sql presence-create.sql Contents of kamdbctl.mysql # path to the database schemas DATA_DIR="/usr/local/share/kamailio/" if [ -d "$DATA_DIR/mysql" ] ; then DB_SCHEMA="$DATA_DIR/mysql" else DB_SCHEMA="./mysql" Fi for TABLE in $STANDARD_MODULES; do mdbg "Creating core table: $TABLE" sql_query $1 < $DB_SCHEMA/$TABLE-create.sql if [ $? -ne 0 ] ; then merr "Creating core tables failed!" exit 1 fi done Kurt Mullen

2 1

is_method with pv argument?
by Juha Heinanen 16 Dec '10

16 Dec '10

is there a means to test method of sip request against a pseudo variable, which would contain an integer bitmap of methods corresponding to enum request_method values? -- juha

2 5

Changing the text of reason header in CANCEL messages
by Klaus Feichtinger 16 Dec '10

16 Dec '10

Hello, I want to ask if you know any possibility in changing the text of the SIP reason header in CANCEL requests. Is the limitation still the same as described in the mail below from May 2010? For my scenario I have to use the TM module, because incoming requests from the PSTN gateway are parallelly forked to two destinations. Whenever the pending call is cancelled by the PSTN gateway the CANCEL request is still containing a reason header (Q.850;cause=16). But I would like to adapt the text of this header according RFC3326 (e.g. 'Reason: SIP ;cause=487 ;text="Request Terminated"'). The feature "e2e-cancel-reason" is working fine - the original text is copied, but no textops function can make any change..... Is there any trick possible to change the text of this header before it is handled by the proxy (e.g. in creating a spirale - change the text and forward it to the same proxy again)? Or in worst case if this is currently absolutely impossible - would this be an acceptable request for a new feature in any upcoming release of Kamailio 3.1.x? In the release 3.1.x it is still possible to store the text of an incoming reason header, so I guess it could be no problem in overwriting this text with any function...... Is anybody else interested in this feature? regards, Klaus ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Am 21.05.2010 09:26, schrieb François BERGANZ: When I use forward, it doesnt work! I don't know how to do It works fine for me: route{ if (is_method("CANCEL")) { if (!is_present_hf("Reason")) { xlog("L_ERR", "reason missing"); append_hf("Reason: Q.850; cause=31\r\n"); xlog("L_ERR", "Reason added"); } forward(); exit; } Note: above code snippet only works if the proxy does not do any parallel forking of the INVITE. It may also be necessary to change the request-URI to forward the request to the same destination as the INVITE. in: U 2010/05/21 10:29:38.372934 83.136.33.3:16534 -> 88.198.53.113:5060 CANCEL sip:01505641...@83.123.45.165 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.51:16534;branch=z9hG4bK-d8754z-7b591160b17cfc57-1---d8754z-;rport To: "01505641636"<sip:01505641...@83.123.45.165> From: <sip:kl...@83.123.45.165>;tag=7c681960 Call-ID: MzcxYTgzZmIwMGYyMTA4YzVkN2IzZDhjYjhjYjYwNTk. CSeq: 2 CANCEL User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 out: U 2010/05/21 10:29:38.373331 88.198.53.113:5060 -> 83.123.45.165:5060 CANCEL sip:01505641...@83.123.45.165 SIP/2.0 Via: SIP/2.0/UDP 88.198.53.113;branch=z9hG4bK-d8754z-7b591160b17cfc57-1---d8754z- Via: SIP/2.0/UDP 10.10.0.51:16534;received=83.136.33.3;branch=z9hG4bK-d8754z-7b591160b17cfc57-1---d8754z-;rport=16534 To: "01505641636"<sip:01505641...@83.123.45.165> From: <sip:kl...@83.123.45.165>;tag=7c681960 Call-ID: MzcxYTgzZmIwMGYyMTA4YzVkN2IzZDhjYjhjYjYwNTk. CSeq: 2 CANCEL User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 Reason: Q.850; cause=31 regards Klaus

2 2

TCP Overload Congestion
by Jijo 16 Dec '10

16 Dec '10

Hi Andrei, I'm observing TCP Recv buffer getting full when we are doing a load run (30cps) on Proxy with TCP. Basically the congestion is happening on the TCP connection from SIP Server to Proxy. I beleive kamailio is not processing the message fast enough, due to this congestion is occuring. Did anyone observe this behavior? How do we resolve it? Thanks Jijo

3 3

← Newer
1
2
3
4
5
6
7
8
9
...
12
Older →

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users December 2010