- sr-users - kamailio.org

Re: [Kamailio-Users] Is Timestamp supported in kamailio?
by Iñaki Baz Castillo 17 Mar '10

17 Mar '10

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi all, > I would like to asking you Hi, it's just better to keep the thread in the maillist. > about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. That's not true. The request can contain a header like: TimeStamp: 25 and the proxy/server replies in the 100 Trying: TimeStamp: 25 0.001 This means that it has token 0.001 seconds to the server in order to process the request and generate the 100 response. -- Iñaki Baz Castillo <ibc(a)aliax.net> -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] [OT] MSN presence flow
by Iñaki Baz Castillo 17 Mar '10

17 Mar '10

Hi, does somebody have a MSN protocol flow related to presence rules or buddies management? This is, I would like to know how MSN protocol imlements some tasks as: - Adding a buddy. - Blocking a buddy for presence. - Blocking a contact (not a buddy) for presence. Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] kamailio crashes
by Panagiotis Skoulikaritis 17 Mar '10

17 Mar '10

Hello Daniel I do have quite a few core files, please send me the gdb commands. Regards Panagiotis. Daniel-Constantin Mierla wrote: > Hello Panagiotis, > > On 03/16/2010 02:21 PM, Panagiotis Skoulikaritis wrote: >> Hello Daniel >> >> I will need time to recreate the problem, >> attached are the only traces I kept. >> on the amaze-4.cap the calls are originated by a softphone registered >> on the kamailio >> on the crash.cap the calls are originated from the "PSTN". >> >> we do account the $pd and the $pn >> >> I reply to you directly since i don't want to give the traces on the >> mailing list, I hope you will understand. > > I asked to be sent private, it is ok. > >> >> Also for workaround Alex took out the PDT module from the route and >> the kamailio do not crash anymore. > > Hmm, so you say pdt module is related? That is pretty small and old > module... > > Do you still have the core file? I can send you some gdb commands to > get more details out of it. > > Thanks, > Daniel > >> >> Regards >> >> Panagiotis >> >> >> Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> can you send me ngrep/pcap file with ip addresses so I can match >>> which 200ok is causing the problem (coming from B or coming from >>> Asterisk)? The backtrace shows ip while the sip trace is masked. >>> >>> Also, I would need a bit more info from the core file. Please keep >>> one around. The issue seems to be related to P-Asserted-Identity >>> header, but I couldn't find any such header in the sip trace you sent. >>> >>> Are you accounting the PAI header? >>> >>> Thanks, >>> Daniel >>> >>> On 03/10/2010 03:30 PM, Panagiotis Skoulikaritis wrote: >>>> Dear Marius >>>> >>>> The scenario is as follows: >>>> >>>> 1. A Call is placed by a sip subscriber "A" >>>> 2. kamailio forwards the call to the asterisk server >>>> 3. Asterisk plays an IVR message on the subscriber "A", creates a >>>> new call to a "virtual" number which is forwarded to the kamailio >>>> server, and plays an ivr to this leg as well when the call is >>>> answered, then it connects the two calls. >>>> 4. Kamailio translates the "virtual" number to the pstn number of >>>> subscriber B >>>> >>>> >>>> I have attached a picture of the above scenario. >>>> >>>> The modules that are loaded are: >>>> >>>> loadmodule "db_mysql.so" >>>> loadmodule "mi_fifo.so" >>>> loadmodule "mi_datagram.so" >>>> loadmodule "sl.so" >>>> loadmodule "tm.so" >>>> loadmodule "rr.so" >>>> loadmodule "pv.so" >>>> loadmodule "maxfwd.so" >>>> loadmodule "usrloc.so" >>>> loadmodule "registrar.so" >>>> loadmodule "textops.so" >>>> loadmodule "uri_db.so" >>>> loadmodule "siputils.so" >>>> loadmodule "xlog.so" >>>> loadmodule "acc.so" >>>> loadmodule "dispatcher.so" >>>> loadmodule "pdt.so" >>>> loadmodule "dialplan.so" >>>> loadmodule "siptrace.so" >>>> loadmodule "dialog.so" >>>> loadmodule "sqlops.so" >>>> loadmodule "userblacklist.so" >>>> loadmodule "htable.so" >>>> loadmodule "uac.so" >>>> >>>> >>>> The config that does all the routing is : >>>> >>>> route[10] { >>>> >>>> xlog("alx ------- This is Route 10 -------"); >>>> >>>> if($rU =~ "^.*%+") >>>> { >>>> xlog("alx ------- The number contains %23 "); >>>> $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if($rU =~ "^.*#+") >>>> { >>>> xlog("alx ------- The number contains #"); >>>> $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if(prefix2domain("2", "0")) { >>>> >>>> $var(dial_grp) = $(rd{s.select,0,.}{s.int}); # >>>> Dialplan group prefix for routing >>>> $var(num_pr) = $(rd{s.select,1,.}{s.int}); # The >>>> number of digits that prefix has >>>> $var(num_translation) = >>>> $(rd{s.select,2,.}{s.int}); # Called number translation >>>> $avp(s:port_translation) = >>>> $(rd{s.select,3,.}{s.int}); # Port number translation >>>> #$var(test_var) = $(rd{s.select,4,.}{s.int}); # >>>> Future property >>>> >>>> $avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)}); >>>> $rU = $(rU{s.substr,$var(num_pr),0}); >>>> >>>> xlog("alx ------- The new rU is $rU and properties $rd >>>> -------"); >>>> >>>> if($var(num_translation) == 1) >>>> { >>>> if($sht(a=>$rU)!=null){ >>>> >>>> $rU = $sht(a=>$rU); >>>> xlog("alx ------- Translation Done. DST >>>> num=$rU ----------"); >>>> >>>> } else { >>>> xlog("alx ------- Translation NOT Done >>>> ----------"); >>>> } >>>> >>>> >>>> #xlog("alx ------- We have DST number >>>> translation for user fU $avp(s:frm_user_name) ----------"); >>>> #if(dp_translate("31", "$rU/$rU")) >>>> #{ >>>> # xlog("alx ------- Translation Done. >>>> DST num=$rU ----------"); >>>> #} else { >>>> # xlog("alx ------- Translation NOT >>>> Done ----------"); >>>> #} >>>> } >>>> >>>> if(dp_translate("$var(dial_grp)", "$rU/$rU")) >>>> { >>>> xlog("alx ------- The $rU and with >>>> attributes :$avp(s:dest) -------\n"); >>>> >>>> $var(i) = 0; >>>> while($(avp(s:dest){s.select,$var(i),.})!="#") >>>> { >>>> $avp(s:dstgrp) = >>>> $(avp(s:dest){s.select,$var(i),.}{s.int}); >>>> $var(i) = $var(i) + 1; >>>> xlog("alx ------- The >>>> avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------"); >>>> } >>>> >>>> # backup the username so we can use >>>> different prefixes >>>> $avp(s:user) = $rU; >>>> >>>> # select destination from first group >>>> >>>> if(ds_select_domain("$avp(s:dstgrp)", "4")) >>>> { >>>> >>>> if($(ru{uri.param,prefix})!=null) >>>> { >>>> $ru >>>> = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; >>>> >>>> } else { >>>> $ru >>>> = "sip:" + $avp(s:user) + "@" + $rd; >>>> } >>>> } >>>> >>>> $avp(s:dstgrp) = null; >>>> xlog("alx ------- The final >>>> RURI is $ru ------- "); >>>> if($avp(s:port_translation) >>>> == 1) >>>> { >>>> rewriteport("5061"); >>>> } >>>> t_on_failure("3"); >>>> t_relay(); >>>> exit; >>>> >>>> } >>>> >>>> >>>> >>>> >>>> } >>>> >>>> >>>> } >>>> >>>> Attached is the trace >>>> >>>> Regards. >>>> >>>> P. >>>> >>>> marius zbihlei wrote: >>>>> Panagiotis Skoulikaritis wrote: >>>>>> Hello Daniel >>>>>> >>>>>> the kamailio version is 1.5.3 >>>>>> >>>>>> Regards >>>>>> >>>>>> P. >>>>> Hello, >>>>> >>>>> Can you give us more details like the sip message that generates >>>>> the coredump (or if every sip message received generates the >>>>> core), if your config does something more out of the >>>>> ordinary(let's say exotic). Can we reproduce it ? >>>>> >>>>> It would also be helpful if you specify the list of modules you >>>>> have loaded. >>>>> >>>>> Cheers, >>>>> Marius >>>>>> >>>>>> Daniel-Constantin Mierla wrote: >>>>>>> Hello, >>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> Daniel-Constantin Mierla >>> Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 >>> * http://www.asipto.com/index.php/sip-router-masterclass/ >>> > > -- > Daniel-Constantin Mierla > Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 > * http://www.asipto.com/index.php/sip-router-masterclass/ >

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Re: [Kamailio-Users] Is Timestamp supported in kamailio?
by Iñaki Baz Castillo 16 Mar '10

16 Mar '10

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi Castillo, > I would like to asking you about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is there any open source support the time stamp header. > > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. > > Please Help me; I think I *already* helped you as you *already* asked me directly. Please, keep the threads in the maillist, nobody likes to give private free support. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Is Timestamp supported in kamailio?
by mustafa rifaee 16 Mar '10

16 Mar '10

Hi all; I would like to ask you about the timestamp in kamailio, Is it supported or not ? if yes how i can enable it ? Please Help me; Thanks. Best Regards Mustafa Rifaee

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[Kamailio-Users] [dialog module] dialog is not terminated when no ACK is received
by Iñaki Baz Castillo 16 Mar '10

16 Mar '10

Hi, using Kamailio 1.5.4. I use dlg_manage() for an INVITE. 200 Ok is replied by the callee but the UAC doesn't send the ACK (due to a crash). The dialog remains in Kamailio dialog memory/table in state 3 and would expire after default_timeout (which usually is 3600 seconds or more, unsuitable for this case). Yes, it could occur that the proxy is not doing loose_routing, but in that case it doesn't make sense to use dialog module, so shouldn't dialog module expire dialogs in state 3 after ~32 seconds? I'm using profiles_with_value to limit the number of calls per user, so this issue is a bit important, as a UAC not sending the ACK for a 200 means one less available channel for this user during dialog module default_timeout. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] maintenance this afternoon
by Daniel-Constantin Mierla 15 Mar '10

15 Mar '10

Hello, some small maintenance tasks will be performed afternoon today to the server hosting the mailing lists and kamailio.org site, if you notice some downtime, try again a bit later. Thanks, Daniel

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[Kamailio-Users] kamailio crashes
by Panagiotis Skoulikaritis 14 Mar '10

14 Mar '10

Dear fellow kaimailio users We have a kamailio server which crashes. below is the backtrace from the core files any idea why the kamailio is crashing Regards Panagiotis core.29568 Mar 10 09:27 #0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 #1 0x0000000000423d5c in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x00000000004790cc in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 core.19350 Mar 10 09:05 (gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x00002ad23fe3e525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad2432dcf58, rpl=0x772d28, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x7ffff9c016bc) at t_lookup.c:964 #5 0x00002ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x7ffff9c017a0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7ffff9c019b8) at main.c:774 (gdb) core.29567 Mar 10 09:27 gdb) backtrace #0 free_to (tb=0x776460) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at parser/hf.c:187 #2 0x00002ad17d15a525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad1805f8f58, rpl=0x771920, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad17d163fa2 in t_check (p_msg=0x771920, param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 #5 0x00002ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771920) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: <si"..., len=919, rcv_info=0x7fff8bf6a9b0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 (gdb) core.5326 Mar 10 09:02 (gdb) backtrace #0 free_to (tb=0x772c48) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2b2578e22560) at parser/hf.c:187 #2 0x00002b257597b525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b2578e213b8, rpl=0x771b50, code=<value optimized out>) at sip_msg.h:54 #3 0x00002b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002b2575984fa2 in t_check (p_msg=0x771b50, param_branch=0x7fff7cdc63ac) at t_lookup.c:964 #5 0x00002b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771b50) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: <s"..., len=918, rcv_info=0x7fff7cdc6490) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 (gdb) quit

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[Kamailio-Users] Loadbalancer/outbound_proxy and ACK problem
by Pavel Miskov 13 Mar '10

13 Mar '10

Hello list, let me first show my scenario: UAC_A ---> LB ---> | | PROXY+REG #1 | or | PROXY+REG #2 UAC_B <--- LB <--- | LB cfg: modparam("path", "use_received", 1) loose_route(); #for REG+PROXY if(is_method("REGISTER")){ add_path_received(); ds_select_dst("1", "0"); fix_nated_register(); } else { record_route(); add_path(); ds_select_dst("1", "0"); fix_nated_register(); } PROXY+REG cfg: ------------- modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) INVITE if working fine until ACK. UAC_A sends ACK to LB "ACK sip:UAC_B@LB_IP:5678;" insted of "ACK sip:UAC_B@UAC_B_PUB_IP:". In "200 OK" which PROXY sends to LB Contact is rewritten with ip addres of LB "Contact: <sip:UAC_B@LB_IP:5678;" and this is why UAC_A sends wrong ACK. Obviously I miss something so if someone could point me where to look. Thanks Pavel U UAC_A_PUB_IP:31488 -> LB_IP:5678 INVITE sip:UAC_B@test.com SIP/2.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport..Max-Forwards: 70..Contact: <sip:UAC_A@UAC_A_PRIV_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID:ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488;received=UACs_PUB_IP..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U LB_IP:5678 -> REG_PROXY_IP:5166 INVITE sip:UAC_B@test.com SIP/2.0..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Max-Forwards: 69..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U REG_PROXY_IP:5166 -> LB_IP:5678 INVITE sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d SIP/2.0..Record-Route: <sip:REG_PROXY_IP:5166;lr=on;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Route: <sip:LB_IP:5678;lr;received="sip:UAC_B_PUB_IP:31468">..Max-Forwards: 68..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bKa1b1.c7ca19a7.0;rport=5166;received=REG_PROXY_IP..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U LB_IP:5678 -> UAC_B_PUB_IP:31468 INVITE sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d SIP/2.0..Record-Route: <sip:REG_PROXY_IP:5166;lr=on;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Max-Forwards: 67..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U UAC_B_PUB_IP:31468 -> LB_IP:5678 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport =5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@UAC_B_PRIV_IP:31 468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41.. Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip: LB_IP:5678;lr=on>..Contact:<sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To:"UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U UAC_B_PUB_IP:31468 -> LB_IP:5678 SIP/2.0 200 OK..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@UAC_B_PRIV_IP:31468;r instance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 200 OK..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact:<sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 200 OK..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 200 OK..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com >;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U UAC_A_PUB_IP:31488 -> LB_IP:5678 ACK sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d SIP/2.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-0472a03b5d1aa106-1---d8754z-;rport..Max-Forwards: 70..Route: <sip:LB_IP:5678;lr>..Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Contact: <sip:UAC_A@UAC_A_PRIV_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 ACK.. # U LB_IP:5678 -> REG_PROXY_IP:5166 ACK sip:REG_PROXY_IP:5166;lr;nat=yes SIP/2.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.2..Via: SIP/2.0/UDP LB_IP:5678;rport=5678;received=LB_IP;branch=z9hG4bKa1b1.46778065.2..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-0472a03b5d1aa106-1---d8754z-;rport=31488..Max-Forwards: 68..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4 YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 ACK..

3 5

[Kamailio-Users] What is the best module for routing INVITE based on DID in R-URI?
by Geoffrey Mina 13 Mar '10

13 Mar '10

Hello, I am putting up a Kamailio server which will do nothing but route INVITE requests from my upstream carrier to individual offices on my side. The office locations will NOT be registered SIP UAs, but other Kamailio proxy servers. What I want to have is a database of DIDs associated with a forwarding IP:Port and/or SRV records. 5555551212 ==> 1.2.3.4:5060 5555551213 ==> 1.2.3.5:5060 etc. Any guidance on what the best approach to achieving my goal would be much appreciated. Thanks, Geoff

4 12

[Kamailio-Users] need help to use the Kamailio register to remote sip proxy
by Cucku Cucku 12 Mar '10

12 Mar '10

Hi all i found the Kamailio supports register to remote sip proxy: http://www.mail-archive.com/users@lists.kamailio.org/msg07451.html it mentions to use : SQL to create the mysql table is in utils/kamctl/mysql/uac-create.sql but i didnt find the sql script My sip proxy version : sercmd> core.version Server: kamailio (3.0.1 (i386/linux)) 679736 my util folder : [root@localhost kamailio-3.0.1]# ls utils/kamctl/mysql/ acc-create.sql domainpolicy-create.sql purple-create.sql alias_db-create.sql drouting-create.sql registrar-create.sql auth_db-create.sql group-create.sql rls-create.sql avpops-create.sql htable-create.sql siptrace-create.sql carrierroute-create.sql imc-create.sql speeddial-create.sql cpl-create.sql lcr-create.sql standard-create.sql dialog-create.sql msilo-create.sql uri_db-create.sql dialplan-create.sql pdt-create.sql userblacklist-create.sql dispatcher-create.sql permissions-create.sql usrloc-create.sql domain-create.sql presence-create.sql please help Thank you Muốn đổi mới cách xưng hô với ông xã/ bà xã của bạn? Xem có bao nhiêu cách xưng hô giữa vợ và chồng nhé! http://vn.answers.yahoo.com

3 3

[Kamailio-Users] Stateless failover
by Santiago Soares 12 Mar '10

12 Mar '10

We're using Kamailio as a stateless load balancer, and would like to add failover support. Is it possible to have failover support in stateless mode? What i want is that when a server doesn't answer to a request, Kamailio set its state to probing, and start to ping it until it answer, then setting its state to active again. And that only active servers are selected by ds_select_dst(). Is it possible? How can I do this? Thank you. Santiago

4 4

[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 12 Mar '10

12 Mar '10

Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

3 2

[Kamailio-Users] IP Authentication
by Joao Gomes Pereira 12 Mar '10

12 Mar '10

Hello to all Is is possible to implement IP authentication in Kamailio? Does Kamailio has a manual to do that? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

5 8

basic sip-router installation tutorial
by Meftah Tayeb 12 Mar '10

12 Mar '10

hi all, please could anyone give me a simple step by step tutorial to setup sip-router in a debian OS with mysql and other things? from jit, no zip file... thanks

4 3

[Kamailio-Users] Handling 3XX messages
by Bruno Machado 11 Mar '10

11 Mar '10

Hi! I have a Kamailio 1.5.2 handling 3XX messages. The commands below are executed in a failure route. If the first contact is not reachable or even a negative answer is received, the algorithm will call a second failure route, executing the 't_next_contacts'. Following this logic, the server will try several endpoints. The first 200 OK received from one this endpoints will establish the call. If all the contacts are tried, the server will finish the session with sending an error to the source. get_redirects("15:15","Redirect"); if(!t_load_contacts()) { t_reply("500", "Server Internal Error - Cannot load contacts"); exit; }; if(!t_next_contacts()) t_reply("404", "Not found"); else t_relay(); Ok... I found just one problem. If the 3XX message brings 5 different contacts, (all them with different q value), Kamailio just try 4 of them. Contact: sip:20182#556250982222@aaa.aaa.aaa.aaa;q=0.015, sip:3441#556250982222@bbb.bbb.bbb.bbb;q=0.014, sip:3441#556250982222@ccc.ccc.ccc.ccc;q=0.013, sip:50014#556250982222@ddd.ddd.ddd.ddd;q=0.012, sip:777#556250982222@eee.eee.eee.eee;q=0.011 After the 300 message, the server sends a invite to the first contact. Seconds later, analysing the debug screen, I see: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@bbb.bbb.bbb.bbb The INVITEs are sent to this IP. After some seconds the server calls the third contact: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@ccc.ccc.ccc.ccc And so on until this, the last contact: DBG:tm:t_next_contacts: next contact is <sip:777#556250982222@eee.eee.eee.eee ERROR:core:add_avp: 0 ID or NULL NAME AVP ERROR:tm:t_next_contacts: setting of fr_inv_timer_avp failed The proxy sent INVITEs for all them, except the last one because this failure. The function 't_next_contact' returns an error value. Somebody has a hint about this? Regards, bruno machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 11 Mar '10

11 Mar '10

**resending as I am not sure if this made it out the first time, I do not believe it did, if this is a duplicate -- my apologies.** Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

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[Kamailio-Users] 400 Bad Request
by Ernest Mavrel 10 Mar '10

10 Mar '10

3 2

[Kamailio-Users] cdr_id not registered in acc
by Anders 10 Mar '10

10 Mar '10

Hi, (Kamailio 1.5 + freeradius + cdrtool) After a call is finished, it's INVITE and BYE are registered just fine in the acc table, but the records in acc never receive a cdr_id, and (therefore) nothing is inserted the cdrs table. Ideas on where to start looking? Thanks, Anders

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[Kamailio-Users] DEFINE question
by Uriel Rozenbaum 10 Mar '10

10 Mar '10

Hi guys, The question's simple, is there any pre-processor command to DEFINE constants? Thanks! Uriel

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[Kamailio-Users] OpenSER+Asterisk+BLF
by Mark Sayer 10 Mar '10

10 Mar '10

We are needing to modify the configure of a currently operating OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that are sent between Asterisk and a phone that supports BLF (like the Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 & Asterisk 1.4.17 in the same box. OpenSER performs all registration, authentication and NAT. Asterisk handles the media and the accounting. In a pure Asterisk environment a "hint" would be setup in the Asterisk extensions.conf file and the phone (UA) would SUBSCRIBE to that HINT. Once Asterisk has registered that UA to the HINT then it sends SIP-NOTIFY messages to the UA as the status of the channel changes (available, ringing, busy). Our current openser.cfg file makes no mention of either SUBSCRIBE or NOTIFY which is an obvious reason that my Asterisk installation never registers the UA to the HINT. Is anyone interested in getting paid to fix this for us (we're too stupid to do it ourselves) or to offer another solution for controlling BLF in this setup. Thanks, Mark

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[Kamailio-Users] small typo in rewriteport example on Kamailio Wiki
by Andreas Heise 09 Mar '10

09 Mar '10

Hello, there is a small typo in rewriteport example on Kamailio Wiki should be rewriteport("5070"); instead of rewriteport(5070); http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#rewriteport_s… regards, Andreas

2 1

[Kamailio-Users] Kamailio (OpenSER) v3.0.1 Released
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, the first patch release for 3.0 series is out as version 3.0.1. It includes the fixes to issues discovered since 3.0.0. Database structure and configuration file compatibility are preserved so the upgrade from 3.0.0 is straightforward. Links and more details are available at: http://www.kamailio.org/w/2010/03/kamailio-v3-0-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] [OT] BYE sip:SIMPLE&XCAP@ietf.org SIP/1.0
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, if somebody is bored and has 5-10 minutes of spare time I suggest to read the mail I've sent to sip-implementors maillist, full of hate and fury: https://lists.cs.columbia.edu/pipermail/sip-implementors/2010- March/024529.html Enjoy. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Serusers] ser numbers going up
by Sabatella, Michael 08 Mar '10

08 Mar '10

Hi all, Running ser 0.9.6. I notice these numbers going up and never going back down: I am a bit concerened with the red text. Is there any reason why that does not come down to 0. Do I need to restart the ser process once a month or something around that timing? So far I have not heard of any issues with user call setup but I just want to remain proactive. Serctl moni outputs the following: Server: Sip EXpress router (0.9.6 (i386/linux)) Now: Mon Mar 8 10:05:18 2010 Up Since: Fri Jan 8 11:24:17 2010 Up time: 5092861 [sec] Transaction Statistics Current: 1 (176 waiting) Total: 5624219 (0 local) Replied localy: 6049354 Completion status 6xx: 0, 5xx: 121, 4xx: 308504, 3xx: 0,2xx: 5316846 Stateless Server Statistics 200: 13145729 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 42 401: 0 403: 0 404: 350766 407: 16536998 408: 0 483: 0 4xx: 2 500: 1 5xx: 0 6xx: 0 xxx: 0 failures: 72 Michael Sabatella IPC Systems Network Services Engineer Office: 1-201-253-2280 Cell: 1-908-380-5321 @ IPCGlobalNetworks.com<http://www.ipcglobalnetworks.com/> ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] Get all the values for dialogs with a specific profile
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, by running "kamctl fifo profile_get_size togw" I can see the number of dialog tagged as "togw". Also if I run "kamctl fifo profile_get_size togw 999888777" I can see the number of dialogs tagged as "togw" with value "999888777". Is there any way to display all current values for dialogs with a specific profile/tag (i.e. "togw")? The idea is building a simple script to display the current number of calls per user (having that the value applied to dialog profiles means the calling user). Thanks for any suggestion. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] new website look
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, http://www.kamailio.org has a new look! The former site was built with a cms system that discontinued the development, providing no longer updates for newer technologies and nicer design. We hope the new design helps to locate project resources easier and makes site navigation more straightforward via comprehensive list of links in the right sidebar. Credits go to Elio Rojano for the logo artwork and to developers of GPL Wordpress theme Cordobo Green Park which was used as start for the new design. If you are skilled in web design and can build a more personalized look on top of wordpress, please contact us. Cheers, Daniel PS. Do not forget, if you are tomorrow in London, UK, consider attending Present and Future of SIP Routing to get latest update about the project: http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/ -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Dispatcher ds_list results in flag=P
by Geoffrey Mina 08 Mar '10

08 Mar '10

Hello, I have a weird issue with Dispatcher module. When I do a ds_list I get some of my destinations returned with flag=P. It appears that when they are in this state, they are not included in the list of destinations to send traffic to. I have tried restarting Kamailio, and these certain servers always return to flag=P. I have confirmed the servers are up and running fine. They are communicating without issue at the SIP and network level. Any idea what could cause this? Thanks,

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Re: [Kamailio-Users] Users Digest, Vol 57, Issue 85
by Zhe 07 Mar '10

07 Mar '10

Help. Please remove me from this mail list. Thanks 在2010-02-27 19:00:01，users-request@lists.kamailio.org 写道： >Send Users mailing list submissions to > users(a)lists.kamailio.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > >You can reach the person managing the list at > users-owner(a)lists.kamailio.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. remove parameter (Elgar Onel) > 2. Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > 3. Re: Exotic case (but real) in which RtpProxy doesn't work > (Daniel-Constantin Mierla) > 4. Re: Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > 5. Re: Exotic case (but real) in which RtpProxy doesn't work > (Daniel-Constantin Mierla) > 6. Re: Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Fri, 26 Feb 2010 03:00:25 -0800 (PST) >From: Elgar Onel <elgaronel(a)yahoo.com> >Subject: [Kamailio-Users] remove parameter >To: users(a)lists.kamailio.org >Message-ID: <444271.31134.qm(a)web114207.mail.gq1.yahoo.com> >Content-Type: text/plain; charset="us-ascii" > >Dear friends, > >how to remove parameters from INVITE sip address? > >tia, e. > > > > >

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[Kamailio-Users] 3.0 Modules
by dotnetdub 07 Mar '10

07 Mar '10

Hi List, I see two modules in particular that I could really do with running on 1.5.3 - I have started testing 3.0 but will not put into production just yet. Call control and topoh Would it be possible to backport these to 1.53 ? Would there be much to it.?. Call Control in particular is something that is very useful for us. We will eventually move to 3.0 of course as it is amazing product with lots of really nice new features just been burned badly before by moving to something so new. TIA. Stephen.

2 1

[Kamailio-Users] 10 years sip express router (ser)
by Daniel-Constantin Mierla 05 Mar '10

05 Mar '10

Hello, yesterday at CeBIT I've run into Fraunhofer booth and a banner attracted my attention, photo at: http://www.asipto.com/gallery/v/cebit2010/P3030721.JPG.html?g2_imageViewsIn… According to the middle right bubble, 2000 was the first year FhG presented SER at CeBIT. Given that, means the project should celebrate (at least) 10 years these days?!? I was among FhG staffers at CeBIT 2002, presenting latest SER at that time. According to CVS repository on BerliOS, files such as Makefile and main.c are dated 2001. Officially, SER was GPL-ed and open sourced in 2002 (year when ser project was registered to berlios.de software forge). Andrei, more insights from non-open-source era? I got into the project in January 2002, would be interesting to trace the birthday... Cheers, Daniel

2 1

[Kamailio-Users] local 408 from fr_timer and fr_inv_timer
by Robert McGilvray 04 Mar '10

04 Mar '10

I'm looking for a way to distinquish between locally generated 408s triggered by the timers. My problem is that I'm using gateway failover in a failure route and on 408s it would retry phones multiple times if they didn't answer (no voicemail etc) - so fr_inv_timer was triggered after receiving a provisional reply. I need to failover in case a gateway doesn't respond with provisional (it's set at 5 seconds currently). Will this work, is there a cleaner solution? I haven't had the oppurtunity to test it so I may have some syntax errors. onreply_route { if (t_check_status("1[0-9][0-9]")) { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu TO $ru -> Provisional Response --> $T_reply_code\n"); setflag(FLAG_PR_RECEIVED); } } failure_route[ROUTE_DP_FAILURE] { if (t_local_replied("last") && t_check_status("408") && isflagset(FLAG_PR_RECEIVED)) { xlog("L_INFO", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu TO $ru -> Local 408 after provisional response, giving up\n"); exit; } } Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Present and Future of SIP Routing 2010 London
by Daniel-Constantin Mierla 04 Mar '10

04 Mar '10

Hello, I am pleased to announce a dedicated event in London, Tuesday, March 9 afternoon, where to present the version 3.0 and latest developments in Kamailio and SIP Router projects. The event includes as well presentations from various players in the VoIP and SIP market, more details at: http://www.kamailio.org/mos/view/Present-and-Future-of-SIP-Routing-2010-Lon… It is a free event, the number of seats is limited, room being available courtesy of Sun Microsystems (address 55 King William St, London EC4), therefore the first come first serve policy is applied. You can also enroll for a presentation, just send a short description about the topic in the registration email. Hope to see many of you in London next week, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Is this a SER Problem?
by Samson Luk 03 Mar '10

03 Mar '10

I use SPA3102 with a iptel.org VoIP a/c, whenever my Firewall external IP changed (SPA3102's WAN port only connected to the internal LAN with a static IP, router function not used) the iptel.org VoIP a/c will report registration failed. I have to remove the proxy domain field and re-insert it then reboot to fix the problem, just wonder if this is a problem on the SER side or on the SPA side. I am sorry if this is not a correct appropriate place to ask such a question.

1 0

Re: [Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

On 03/02/2010 11:25 AM, Even André Fiskvik wrote: > Thanks for the answers Daniel. > > On 1. mars 2010, at 12.23, Daniel-Constantin Mierla wrote: > > >> Hello, >> >> On 02/28/2010 06:43 PM, Even André Fiskvik wrote: >> >>> Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. >>> In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? >>> >>> >> yes and 'no' :-). Yes because nothing was done in this respect to the module itself. 'No' because now (kamailio 3.0) you can use transformations to get content of xml documents and if the body is xhtml. >> > That's good to know. > > >>> If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? >>> As to my knowledge it can only send text/html ? >>> >>> >> You can build a new message with uac module or update the existing SIP message (content type, body), do msg_apply_changes(): >> http://kamailio.org/docs/modules/3.0.x/modules_k/textops.html#id2513318 >> >> and they use xmpp functions. >> > I did a "quick" hack to support relying the MESSAGE by just setting the content-type to text/plain for now to do some testing. > The error about unsupported content-type thus went away. > > >> You can use ngrep to sniff the network to see connection between kamailio (openser) and xmpp server as well as run kamailio in debug mode in order to get more verbose log messages for each operation. >> > I've tried sniffing traffic on the server, and it doesn't seem like kamailio tries to connect to the XMPP server at all :/ > Any clues what might be causing this? Did you run in debug mode? Can you post the log messages? > I've attached my kamailio config-file as well if anyone would take You sent the email only to me. Did you intend it for mailing list? I recommend to keep sending to mailing list apart of messages with private data. In this way many can reply and other can learn from discussion. > a look at it (the main config is in xmpp.cfg) and also a diagram showing how the components are realated to each other. > I will try to look over configs when I get some time -- because of a trip ahead I am pretty constrained today. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] preparing for kamailio 3.0.1
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

Hello, I think it is time to prepare for first minor release (3.0.1) in 3.0 series. We are approaching two months since kamailio 3.0.0, period within several issues were reported and fixed. Please report anything you are aware is not currently working in 3.0 to sr-dev(a)lists.sip-router.org or bug tracker: http://sip-router.org/tracker/ As for date, end of this week or beginning of next week would be a good time? Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Need a way to substitute SIP Response Code 504 for something else
by Frank Durda IV 03 Mar '10

03 Mar '10

There is a commercial switch vendor who has recently decided to make two changes that combined are quite evil: 1, They decided to devivate from RFC 3398 and when a given call INVITE timer expire (any of the SS7 ones, plus the SIP ones, including any limit specified in the INVITE itself), instead of translating the ISUP Release Cause 102 into a SIP Response Code of 504 (Gateway Timeout), they send out a 503 (Service Unavailable) instead. (503 is commonly used in low-cost carriers and the commercial equipment they use to trigger a route-advance, where the call doesn't end but is relaunched via another slightly more expensive route/carrier who perhaps can get the call to the destination. Some of these low-cost outfits may retry the same call with five or six carriers before reluctantly sending to the local access tandem which is usually the most expensive choice, but almost certain to work. This probably isn't what the inventors of the 503 code had in mind, but that is how it is used. A minority use 404 the same way.) 2. The same vendor implemented an automatic "Address Reach" testing mechanism for outbound calls. This thing means that when sending a call to a distant switch, if a single or small number of calls are returned from that distant switch with a SIP Response Code of 504 from the destination, the sending switch will take that destination switch completely out of route ("blacklist") and not send any calls at all to that destination until the blacklist is manually reset. So a result that applies to one call is fatal to all subsequent calls. This also means that if you sent the INVITE Expires timer to a value less than the number of seconds the calling party is willing to let the called number ring (called party doesn't have voice mail or it doesn't pick up for several rings), you trigger a 504 and bingo, you have killed the route between those two switches. So your own choices for how long to wait on ring-back for a call dictate how quickly it could kill the entire route for that call. Now, the two changes taken together are also somewhat convienent because this equipment maker just happened to make it impossible for their own equipment to ever return a 504 and do can never trip off their own booby-trap. So in a way what these two changes effectively create a not-our-hardware-detector, triggering route failures only when in contact with RFC-compliant equipment made by somebody else. I think that behavior is anti-competitive, and probably illegal in quite a few countries (notably the EU), but that is what exists at the moment. This creation started getting deployed in the past few months. So, to get around this nonsense, I need a way in SER to detect that a 504 has come in and change it to something else before sending it on. I won't quibble about the "proxies can't forward a 503", so I won't try that. I would like to turn the 504 into a 404, which is better than having route get pulled all over the place. Now, the last time (a year or so ago) I asked about trying to change a SIP Response Code in SER, the suggested ser.cfg "fix" created all sorts of strange warnings in the logs about timers expiring and I was told to not worry about that. Well, it seemed to cause a lot more problems than just warnings like memory leaks or cores caused by something, so I ended up having to hard-code the translation I wanted into the SER source code and not use rules in ser.cfg to do this. I'm writing now to (A) alert people about what this major brand switch maker is doing with regard to 504s so you will recognize it if you encounter it, but also (B) to see if maybe has a more elegant way to change SIP Responses via the ser.cfg file has emerged. The obvious things like subst subst_uri don't appear to work, obviously busily editing the non-existent INVITE method message and ignoring the reply message. Using a reply/drop combination to generate a new reply and kill the real one doesn't seem to work either. As I recall it ends up sending the response specified in the reply, and then passing the original response as well. "drop" apparently doesn't kill it completely and I suspect reply is really meant to be used only during handling of the method messages, and not the response messages. Suggestions appreciated.

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[Kamailio-Users] kam 1.5/asterisk 1.6 hints/blf?
by Robert McGilvray 02 Mar '10

02 Mar '10

Please excuse the noob questions, I'm not very familiar with presence. Users are registered to kamailio, authenticated against radius with LDAP backend. I'd like to setup some queues on asterisk for helpdesk call center but in order to do it correctly asterisk needs the device state (ringing, in use etc). I assume I'll need to use realtime ldap and setup device entries for all my phones, then use hints on the extensions as if they were local. If I understand correctly this will make asterisk subscribe to kamailio (I will use kams IP in host field), kam then has the responsibility of handling the subscribe/publish/notify stuff. What needs to be done on the kam side? Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Kamailio (1.5 rev5834) and/or rtpproxy gets frozen
by Iñaki Baz Castillo 01 Mar '10

01 Mar '10

Hi, I'm experimenting an annoying issue in a Debian Lenny 32 bits under a DELL 850 (2 cores). Same occured before with Debian Etch 32 and 64 bits in other host (also DELL 850). Kamailio (1.5 rev 5834) behaves as a load balancer in front of 2 PSTN gateways. There are ~200 calls and for all of them rtpproxy is applied. Sometimes, with no reason (not just in the moments of highest traffic) kamailio gets totally frozen, this is, it doesn't reply to SIP messages, neither relays them. Then Kamailio cannot be killed (just with -9). Sometimes it occurs when performing a fifo command (i.e. "kamctl fifo address_reload"). After killing it, kamailio cannot be started again. It starts and logs the usual logs (to syslog) but after "rtp proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled" there is no more logs. Then if I do a "ps aux | grep kamailio" I just see a single process, it's like the children fail when connecting to rtpproxy and die but the master process doesn't realize of it and remains alive showing no error at all. So I suspect from rtpproxy as even when kamailio was killed rtpproxy process consumes lot of CPU. I'll try with rtpproxy-1.2.1 (the latest stable version) as before I was using a old "truk" version (unfortunatelly I've no idea about how to know the CVS revision of a CVS working directory...). Anyhow I also suspect that the problem could be in the server itself, as sometimes the above problem occurs when reloading iptables and so on, very very strange, no a definitive cause or reason, very annoying. So nothing is clear for me. But I know that the problem occurs: - In two DELL 850, one with 32 and the other with 64 bits. - In both Debian Leeny and Etch. - With kamailio 1.5 rev 5834. - With rtpproxy rev XXX (how to get the last modification date in a CVS working dirrectory?) So, the fact is that I'm completely lost, no idea of what is happening. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Project Update, London Meeting, March 9
by Daniel-Constantin Mierla 01 Mar '10

01 Mar '10

Hello, Tuesday, March 9, I am going to be in London and plan to organize a social networking event for folks around SIP/VoIP, Kamailio (OpenSER) & SIP Router. One is going to be the usual evening dinner/pub meeting, see pics from some past events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ I addition, if there are people interested, I would like to have a session in the afternoon (2-3 hours) to give an update about Kamailio (OpenSER) 3.0, present and future development of SIP Router. Time is rather short, but I encourage attendees to prepare short presentations as well, about their products and services, to have more information exchanged there. Let me know if you wish to attend such event in order to find a proper place to host it (drop me an email). Participation is free for everybody, at the dinner, the classic rule, everyone is going to pay for himself/herself. Looking forward to meeting many of you in London. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Even André Fiskvik 01 Mar '10

01 Mar '10

Hi. I've just started looking at kamailio, and I'm experimenting a bit with configuring a SIP <-> XMPP gateway with it using the pua_xmpp module. The configuration part hasn't beed straight-forward, but I think I'm starting to get somewhere. Both kamailio and XMPP/Jabber are new technologies for me serverside, so I don't have a great deal of experience with either. In short, what I'm trying to accomplish at this stage is: - Have CounterPath eyeBeam/Bria to register to my kamailio server - Have Aastra 6755i register to my kamailio server - Connect an XMPP client to my XMPP server - The CounterPath eyeBeam/Bria, Aastra and the XMPP client should all be able to see each others presence - The CounterPath eyeBeam/Bria should be able to IM other eyeBeams/Brias - The CounterPath eyeBeam/bria should be able to IM the XMPP clients and the other way around Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? As to my knowledge it can only send text/html ? Could anyone with a good knowledge of Kamailio and the various modules pua_xmpp, xmpp, pua, pua_bla comment on this kind of setup is at all possible with kamailio and it's current state? Is there any way I can verify that opensips connects to the XMPP server correctly? I tried connecting FreeSWITCH to my XMPP server (Openfire) in component mode, and it would show up connected in Openfire's administration GUI, however I can not see the same when I try to configure kamailio. If anyone has experience with this setup and have another XMPP server they have tested it with, I would also want to try to swap Openfire with that (given they also have some guidance on XMPP server configuration). Best regards, Even André Fiskvik

2 1

[Kamailio-Users] remove parameter
by Elgar Onel 26 Feb '10

26 Feb '10

Dear friends, how to remove parameters from INVITE sip address? tia, e.

1 0

[Kamailio-Users] ratelimit enhancements
by Daniel-Constantin Mierla 26 Feb '10

26 Feb '10

Hi Marius, since you did some updates to this module, I am opening for debate some needed enhancements I did during 3.0 testing phase and want to get opinions how to get in the code repo. Practically is a new module I named for now ratelimit2 and my last idea is to get it named pipelimit in the trunk. The reason for a new module are some major changes. The module uses the same algorithm but its core is overhaul. - definitions of pipes are loaded from database - there can be unlimited number of pipes - pipes are identified by string names - should be possible to reload pipes at runtime (iirc, not yet in) - new pipes can be added at runtime - functions accept variables to identify the pipe Since I never used queues from this module and haven't spent time to understand the concept behind, this functionality is completely missing. The old module might be good to keep in place, probably many people are using it in this form. So, proposals? What is the way to go on? Common code (algorithms) can be made lib at some point. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] tracking new features
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, a reminder for developers to write a short note for each new feature either to NEWS file in source tree or to wiki page: http://sip-router.org/wiki/features/new-in-devel Makes life much easier when the next major release is going to happen and help people to spot quickly what new features are already available in devel branch, hopefully attracting some of them to go for it and test. Now the wiki page is updated with most of NEWS content for 3.1. Thanks, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] CeBIT 2010
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, I am going to visit CeBIT 2010 in Hannover next week. Drop me an email if you are around and want to meet and chat about Kamailio, SIP Router projects, etc. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Storing nonscalar types (lists)
by Alex Balashov 24 Feb '10

24 Feb '10

Are AVP arrays still the only way to create a list of scalars as of K 3.0.0? Or are there other, more graceful ways to accommodate this need now? AVPs are a very useful feature, though the syntax is not documented in 'avpops' or the cookbook except rather incidentally: $var(i) = 0; while(is_avp_set("$(avp(s:somename)[$var(i)])")) { xlog("L_INFO", "Value at index $var(i): $(avp(s:somename)][$var(i)])\n"); $var(i) = $var(i) + 1; } Is there a better way to handle this now, perhaps via some features imported from SER? Script variables ($var(...)) do not support subscripts judging by the cookbook; do they? Are there other options I am unaware of? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

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[Kamailio-Users] Logging of db_extra parameters + serial forking
by Alex Balashov 24 Feb '10

24 Feb '10

Here is a problem I am sure is not new (1.5.x): modparam("acc", "db_extra", "gateway_id=$avp(s:gateway_id)") route { ... $avp(s:gateway_id) = 13; $ru = "some.gw1"; t_on_failure("1"); if(!t_relay()) sl_reply_error(); } # Serial forking rollover route failure_route[1] { if(t_was_cancelled()) exit; $avp(s:gateway_id) = 14; $rd = "some.gw2"; append_branch(); t_relay(); } The problem is, when serial forking happens, the CDR rows in 'acc' still show a value of 13 for 'gateway_id'. This is obviously because the value is sampled at the time of the initial stateful relay processing and is not updated afterward. Any easy/elegant fix? I can think of several fixes that don't involve using 'acc', obviously, but I'd rather avoid that... -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

1 1

Re: [SR-Users] function calling in the config script with $var() as attribute(s)
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Klaus, Alex, you're right. I've seen Klaus's suggestion only after writing my email message. This is the best solution and solves my problem! regards, Klaus > > > Am 24.02.2010 15:41, schrieb Klaus Feichtinger: > > Hello SR-users and -developers, > > > > I have the question if parameters for a function in the config script > > (e.g. "t_relay(host, port)") in general can be set with a variable > > ($var(xy)). Because of having a set of SIP servers / gateways > distributed > > over a wide area and therefore keeping the configuration as simple as > > possible, I would like to define a custom variable, assign it to a $var > > and use this variable as attribute for a relay function. In this case > the > > variable would represent a string of an IP-address (e.g. "10.1.1.55"). > > When this would work I only have to adapt the custom variables on top of > > each config script and could let the rest "as it is". > > > > > > First practical tests have shown, that the functions are NOT happy with > these "attributes". It can be used without any problems for if-else- > > statements, but I fear not for functions.... > > > > The practical scenario looks like: > > > > tc.ip11 = "10.16.65.101" desc ""; > > [...] > > route { > > $var(test) = $sel(cfg_get.tc.ip11); > > [...] > > if (!t_relay($var(test), "5061")) > > A workaround for t_relay: > > $du = "sip:"+$var(test)+":5061"; > t_relay(); > > > klaus > > > > [...] > > } > > > > This configuration resulted in following error messages: > > > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: syntax error > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 38-41: Function parameter with integer > value not allowed#012 > > > > > > Please, could anybody clarify if this should work or not? If not - is > any > > alternative possibility available for "automating" function attributes > > instead of re-writing each function using the same string attribute? > > > > Thanks in advance, > > > > Klaus > > -- GMX DSL: Internet, Telefon und Entertainment für nur 19,99 EUR/mtl.! http://portal.gmx.net/de/go/dsl02

1 0

function calling in the config script with $var() as attribute(s)
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Hello SR-users and -developers, I have the question if parameters for a function in the config script (e.g. "t_relay(host, port)") in general can be set with a variable ($var(xy)). Because of having a set of SIP servers / gateways distributed over a wide area and therefore keeping the configuration as simple as possible, I would like to define a custom variable, assign it to a $var and use this variable as attribute for a relay function. In this case the variable would represent a string of an IP-address (e.g. "10.1.1.55"). When this would work I only have to adapt the custom variables on top of each config script and could let the rest "as it is". First practical tests have shown, that the functions are NOT happy with these "attributes". It can be used without any problems for if-else- statements, but I fear not for functions.... The practical scenario looks like: tc.ip11 = "10.16.65.101" desc ""; [...] route { $var(test) = $sel(cfg_get.tc.ip11); [...] if (!t_relay($var(test), "5061")) [...] } This configuration resulted in following error messages: <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: syntax error <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 38-41: Function parameter with integer value not allowed#012 Please, could anybody clarify if this should work or not? If not - is any alternative possibility available for "automating" function attributes instead of re-writing each function using the same string attribute? Thanks in advance, Klaus -- Sicherer, schneller und einfacher. Die aktuellen Internet-Browser - jetzt kostenlos herunterladen! http://portal.gmx.net/de/go/chbrowser

3 4

[Kamailio-Users] Kamailio 3.0.0: configuration request regarding VARIABLES usage (custom variables + $var(x))
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Hello Kamailio-3 users, I have two questions regarding usage of variables within the config file (as published in the NEW FEATURES of Kamailio 3.0.0). The reason why I want to use variables instead of standard configuration lines is the maintainability of my SIP servers. I have a group of about 20 sip servers distributed over a wide area. Especially routing rules might change during operation of the servers and must be adapted on all servers. Therefore I want to use variables, defined at the beginning of each config file. Within the config file / the different routing blocks I want to reference to these variables. That would be much easier for automatic config script adaptation with a linux script (e.g.) than parsing the whole config file for special text phrases. QUESTION 1: ========== Is it impossible using a variable for requesting a SRC_IP address? With my current configuration I always see an entry in the SYSLOG that looks as follows: "<core> [route.c:1531]: BUG: comp_ip: invalid type for src_ip or dst_ip (20)" The configuration lines look like: $var(lst_gwIpRange) = "10.16.48.6[0-2]"; [...] if (src_ip=~$var(lst_gwIpRange)) { [...] } Is maybe a transformation missing/necessary to avoid the error message (which causes a malfunction)? Can anybody give me a hint? QUESTION 2: ========== The second problem I found is, that the config variables can be configured within a route block only and not - as expected - e.g. at the top of the config file (in the same way as the custom variables). I can't use the custom variables as they are, because I will use more complex variables, which are a mixture of up to three custom variables. These "complex" variables will be used in more than one routing block. Therefore I have to define these variables in every routing block...... Is an alternative (easier) solution with a single definition available, too? An example of my problem is as follows: # custom variables defined on top of the kamailio.cfg file tst.loc_ipaddr = "10.10.1.1" desc "IP addr of the local SIP Server" tst.loc_prefix = "11" desc "prefix of this SRV" tst.distRu1 = "51[0-9].*" desc "variable representing ID1" # the final variable should look like "sip:1151[0-9].*@10.10.1.1" # and is used in 3 routing blocks route[] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] route[TEST1] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] route[TEST2] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] Thanks in advance for any hints! Regards, Klaus Feichtinger -- Sicherer, schneller und einfacher. Die aktuellen Internet-Browser - jetzt kostenlos herunterladen! http://portal.gmx.net/de/go/atbrowser

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[Kamailio-Users] 3.0.0 and snmpstats
by Kelvin Chua 24 Feb '10

24 Feb '10

hi guys, anybody here tried using snmpstats with 3.0.0? i keep on getting "No such Object" already verified that kamailio is talking to agentx master. net-snmp version 5.3.3 compiled from source also disabled openssl and md5 in net-snmp to prevent any undefined symbols been reading the mailing lists and so far nothing helped. seems like only a couple of guys are using this module? Kelvin Chua

2 4

[Kamailio-Users] Freeing sqlops results
by Alex Balashov 23 Feb '10

23 Feb '10

Should sql_result_free() always be called, no matter what the circumstance, in order to discharge memory allocated to query result sets and/or metadata? Or is it not necessary in certain situations, for example when script is broken via 'exit', i.e. sql_query("db", "SELECT ...", "res"); if(! $dbr(res=>rows)) { # Is it necessary to call sql_result_free() here? sl_send_reply("404", "Not Found"); exit; } Are results implicitly freed at the end of route script execution anyway? Are they tied to transaction persistence in any way? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

3 3

[Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio for termination
by Slot Zero 23 Feb '10

23 Feb '10

Hello, I am a Kamailio noob :). I am trying to get Asterisk to forward calls to my SIP provider via Kamailio. The same machine is running Kamailio and Asterisk. I do not want to consume credentials as they have to be passed on all the way to my SIP provider. There is no NAT of any sorts. SIP Phone/Users connect to Asterisk.I do not need to authenticate when forwarding call from Asterisk to Kamailio as they are both running on the same server but I do need to make sure that Kamailio dials and forwards 011+number to be sent from local host port 5062(Asterisk listener) to SIP provider only. I have 6 Public IP addresses mapped on the server. I want to use the force_send_socket to allow me to change source IP of SIP requests when being sent to the SIP provider on the basis of credentials username in the request. I have pasted my config below. Please tell me what am I doing wrong here. In the kamctlrc file i have SIP_DOMAIN=localhost Thank you #------CONFIG BEGINS------------------ mpath="/lib/kamailio/modules_k/" debug=3 fork=yes children=4 auto_aliases=no alias=localhost alias=192.168.10.1 alias=192.168.10.2 alias=192.168.10.3 alias=192.168.10.4 alias=192.168.10.5 alias=192.168.10.6 disable_tcp=yes loadmodule "sl.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "/lib/kamailio/modules/tm.so" loadmodule "textops.so" modparam("rr", "enable_full_lr", 1) route { # Sanity Check # ------------ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if(msg:len>2048) { sl_send_reply("413", "message too large to be forwarded over UDP without fragmentation"); exit; }; # Record Route and NAT Preset # -------------------- if (method != "REGISTER") { record_route(); }; # Loose Route # ----------- if (loose_route()) { route(1); return; }; # Call Type Processing # -------------------- if (uri != myself) { route(1); return; }; if (uri == myself) { if (method == "BYE") { route(4); return; } else if (method == "CANCEL") { route(4); return; } else if (method == "INVITE") { route(3); return; } else if (method == "NOTIFY") { sl_send_reply("200", "Understood"); return; } else if (method == "OPTIONS") { sl_send_reply("200", "Got it"); return; } }; route(1); } # Default Message Handling # ----------------------- route[1] { t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } # INVITE Message Handling # ---------------------------------- # ---------------------------------- route[3] { if (uri =~ "^sip:011[0-9]@*") { rewritehostport("sip.voipprovider.com:5060"); if (search("^(Contact|m): .*user01*(a)(127\.0\.0\.1|localhost)")) { force_send_socket(192.168.0.2:5060); }; route(1); return; }; } # CANCEL and BYE Message Handling # ---------------------------------- route[4] { route(1); }

3 8

[Kamailio-Users] Rtpproxy hook
by Vikram Ragukumar 22 Feb '10

22 Feb '10

Hello, I am trying to encrypt/decrypt all packets belonging to an RTP stream. I am using Kamailio v1.5 with rtpproxy, where Kamailio forces the rtpstream through rtpproxy. Does rtpproxy provide a hook so that i can gain access to each RTP packet to perform encryption/decryption ? I do not want to end up having a patch for rtpproxy that i need to update whenever there is a new release of rtpproxy. -- Thanks and Regards, Vikram Ragukumar.

2 1

[Kamailio-Users] Error in Registration with TLS on
by Hemanshu Patel 22 Feb '10

22 Feb '10

Dear friends, Since last few days i am working on Kamailio with TLS support. I had followed each and every steps in installation docs...created certificates as well. Then i started testing the server with TLS on using SIPP. First i didnt added any certificate to SIPP, and Registration wasnt happening... When i added a certificate and key to SIPP....it started working fine....i was been able to test Registrations Successfully. Then i started working with one open source soft phone supporting TLS named mumble. IT Supports. Now i hadnt added any certificate to Mumblem. In my settings of kamailio i have set clietn_verify = 0 and require_client_certificate = 0. So without certificate as well i should be able to Authenticate my self successfully. Instead it gives following error in kamailio log: Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.16.218 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_new: on port 58125, type 3 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: looking up socket based TLS server domain [172.16.16.218:5091] Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_find_server_domain: socket based TLS server domain found Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: found socket based TLS server domain [172.16.16.218:5091] Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_add: hashes: 929, 1 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:handle_new_connect: new connection: 0x7fd6f4a58208 23 flags: 0002 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:send2child: to tcp child 0 0(3296), 0x7fd6f4a58208 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:handle_io: received n=8 con=0x7fd6f4a58208, fd=18 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_add: io_watch_add(0x73a0a0, 18, 2, 0x7fd6f4a58208), fd_no=1 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:tls_update_fd: New fd is 18 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: ERROR:core:tls_accept: SSL_accept failed: SSL_ERROR_SSL Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_del: io_watch_del (0x73a0a0, 18, -1, 0x10) fd_no=2 called Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: releasing con 0x7fd6f4a58208, state -2, fd=18, id=1 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: extra_data 0x7fd6f4a683a0 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:handle_tcp_child: reader response= 7fd6f4a58208, -2 from 0 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_destroy: destroying connection 0x7fd6f4a58208, flags 0002 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_close: closing SSL connection Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_update_fd: New fd is 23 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_shutdown: shutdown successful Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_clean: Cleanup function entered Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.16.218 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_new: on port 58126, type 3 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: looking up socket based TLS server domain [172.16.16.218:5091] Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_find_server_domain: socket based TLS server domain found Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: found socket based TLS server domain [172.16.16.218:5091] Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_add: hashes: 930, 2 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:handle_new_connect: new connection: 0x7fd6f4a58208 23 flags: 0002 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:send2child: to tcp child 0 0(3296), 0x7fd6f4a58208 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:handle_io: received n=8 con=0x7fd6f4a58208, fd=18 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_add: io_watch_add(0x73a0a0, 18, 2, 0x7fd6f4a58208), fd_no=1 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:tls_update_fd: New fd is 18 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: ERROR:core:tls_accept: SSL_accept failed: SSL_ERROR_SSL Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_del: io_watch_del (0x73a0a0, 18, -1, 0x10) fd_no=2 called Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: releasing con 0x7fd6f4a58208, state -2, fd=18, id=2 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: extra_data 0x7fd6f4a683a0 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:handle_tcp_child: reader response= 7fd6f4a58208, -2 from 0 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_destroy: destroying connection 0x7fd6f4a58208, flags 0002 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_close: closing SSL connection Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_update_fd: New fd is 23 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_shutdown: shutdown successful Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_clean: Cleanup function entered And in Mumble soft phone log it gives me following Error: [9:50 AM] Welcome to Mumble. [9:50 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. Can any one suggest what could be the problem? My Server works great with SIPP with TLS....so i dont think theres any config related error and i have set client_require_certificate = 0 thats for sure.... In real life scenario, hard or soft phones wont have certificates...so they should be able to connect to server and authenticate/Authorize themselves if server has valid certificate.But its not happening. So i need help from experienced guys.... -- Regards, Hemanshu Patel M: 09601295238

2 2

[Kamailio-Users] acc
by Anders 22 Feb '10

22 Feb '10

Hi, Probably a simpel fix some...not for me ;-) I get this here below in the log when I try to start Kamailio on a new installation. I suspect something is wrong with the acc module. What I read elsewhere is that it needs to recompile... - could that be, and if so, how would I recompile only a module? Thanks!! //Anders Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_config> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 140, column 25-26: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_flag> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 141, column 27-28: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 142, column 20-21: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_extra> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 157, column 2-3: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:main: bad config file (4 errors)

2 1

[Kamailio-Users] Demerits of using normal Authentication insted of radius or diameter auth
by Hemanshu Patel 21 Feb '10

21 Feb '10

Hello guys..... Recent now i had started taking keen interest in kamailio. I was testing kamailio, and i had test it under two scenario. first of all let me tell u about my configuration. I am testing on normal Desktop pc, installed kamailio,mysql, radius everything on same pc. I am using MYSQL anyway for both sort of testing. When i test with radius authentication, means using radius_www_authorize at max i am getting around 500 cps for Registration, but when i use normal www_authorize i am getting around 2000 CPS.... whopping difference guys..... So, i was wondering what is merits and demerits of not using Radius based authentication. Is it insecure if i use normal Authentication? Its not in my knowledge that radius do encryption...so i dont think by using radius it provides more security. What is your opinions about it? -- Regards, Hemanshu Patel M: 09601295238

3 3

[Kamailio-Users] Username of P-Asserted-Identity header
by Marco Bungalski 19 Feb '10

19 Feb '10

Hi, all! Is there an easy possibility to get the Username of the P-Asserted-Identity header? When i use $ai i get for example "sip:+4942317769999@testdomain.de". I need only "+4942317769999" in a variable. I know for example that i can use $fU and get only the Username (and $fu gives me the whole From URI).... Cheers, Marco Marco Bungalski GmbH Traversale 5 27283 Verden Telefon: +49 4231 - 776 9999 Fax: +49 4231 - 776 9998 Mobil: +49 172 4204774 e-mail: Marco(a)Bungalski.de www.bungalski.de www.t-m-net.de Geschäftsführender Gesellschafter: Marco Bungalski Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586

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[Kamailio-Users] CANCEL goes bananas!
by Anders 19 Feb '10

19 Feb '10

Ok, "bananas" might not be the right word, but it resembles what is happening ;-) When I get a CANCEL from my customer to Kamailio, it is not passed on to my vendor - this is with a specific customer using VoipSwitch. The result is that the phone called keeps ringing even though it's been (attempted) cancelled, and a sideeffect is that it goes nuts at the carrier side because it can send anymore and returns 480 or 486. Below is the dialog from the Session Progress to when the call is cancelled - and then it goes bananas. Any ideas? Thanks a lot! U 2010/02/12 14:40:25.435793 MyKamailioIP:5060 -> MyCustomerIP:5060 SIP/2.0 183 Session Progress. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Contact: <sip:84370551184517324@MyVendorIP:5060>. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. CSeq: 1 INVITE. Content-Type: application/sdp. Via: SIP/2.0/UDP MyCustomerIP:5060;rport=5060;received=MyCustomerIP;branch=z9hG4bk120246101250121320412156. Content-Length: 210. . v=0. o=prxams02 0 0 IN IP4 MyVendorIP. s=sip call. c=IN IP4 62.93.140.13. t=0 0. m=audio 42972 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2010/02/12 14:40:30.625842 MyCustomerIP:5060 -> MyKamailioIP:5060 CANCEL sip:890912551184517324@MyKamailioIP SIP/2.0. CSeq: 1 CANCEL. Via: SIP/2.0/UDP MyCustomerIP:5060;branch=z9hG4bk120246101250121320412156. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. Contact: <sip:MyCustomerIP:5060;transport=udp>. Content-Length: 0. . U 2010/02/12 14:41:25.496605 MyVendorIP:5060 -> MyKamailioIP:5060 SIP/2.0 480 Temporarily Unavailable. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Contact: <sip:84370551184517324@MyVendorIP:5060>. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. CSeq: 1 INVITE. Via: SIP/2.0/UDP MyKamailioIP;branch=z9hG4bKf18c.fa7b81d2.0. Via: SIP/2.0/UDP MyCustomerIP:5060;rport=5060;received=MyCustomerIP;branch=z9hG4bk120246101250121320412156. Content-Length: 0. . U 2010/02/12 14:41:25.496888 MyKamailioIP:5060 -> MyVendorIP:5060 ACK sip:84370551184517324@MyVendorIP:5060 SIP/2.0. Via: SIP/2.0/UDP MyKamailioIP;branch=z9hG4bKf18c.fa7b81d2.0. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. CSeq: 1 ACK. Max-Forwards: 70. User-Agent: Kamailio. Content-Length: 0.

2 3

[Kamailio-Users] Testing of Kamailio TLS with Sipp TLS
by Hemanshu Patel 19 Feb '10

19 Feb '10

Hello friendsm, I am testing TLS feature of both kamailio and Sipp. I first downloaded kamailio 1.5.x TLS supported version, uncommented TLS=1 from Makefiel and then build the kamailio. first i test kamailio without TLS with sipp for registration and everything works file. Then i follow "http://www.kamailio.org/docs/tls-devel.html" and creates rootCA, user certificates and all configuration parameters to kamailio.cfg file Config paras are as below: /* uncomment the following lines to enable TLS support (default off) */ disable_tls = no listen = tls:172.16.16.218:5091 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-calist.pem" and restart kamailio server It works and i can see via netstat on port 5091. but when i starts sipp it gives me following error. [hemanshu@localhost sipp.3.1]$ ./sipp -sf ./data/rauth.xml -inf ./data/user.csv -r 1 -m 1 -trace_err -trace_stat -nd -fd 1 -i 172.16.16.218 172.16.16.218:5091 -t l1 2010-02-18 13:51:40:244 1266481300.244432: FI_init_ssl_context: SSL_CTX_use_certificate_file failed. I know i have built sipp with TLS support, then i can not figure out where is the problem. Sipp says user certification file failed ..but client doesnt need any certification file...or does it? i even tried with tls_verify_client = 1 , tls_require_client_certificate = 1, playing with different combinations...but still same ans. Can anyone suggest me what could be wrong? Have i made any mistake in configuring kamailio or theres some problem in SIPP. -- Regards, Hemanshu Patel M: 09601295238

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[Kamailio-Users] Issue with dialog module
by Santiago Gimeno 19 Feb '10

19 Feb '10

Hi, I'm using kamailio 1.5 from the branch. In a conditional forwarding scenario with this call flow: Phone A Kamailio Phone B Phone C | | | | | | | | | | | | |INVITE | | | |------------->| | | |100 Trying | | | |<-------------| | | | |INVITE | | | |------------->| | | |100 trying | | | |<-------------| | | |180 Ringing | | | |<-------------| | |180 Ringing | | | |<-------------| | | | |486 Busy Here | | | |<-------------| | | |ACK | | | |------------->| | | |INVITE | | | |---------------------------->| | |100 trying | | | |<----------------------------| | |180 Ringing | | | |<----------------------------| |180 Ringing | | | |<-------------| | | | |200 OK | | | |<----------------------------| |200 OK | | | |<-------------| | | |ACK | | | |------------->| | | | |ACK | | | |---------------------------->| | | | | | | | | What I'm observing is that when Kamailio receives the 486 from B, the dialog between A and B is not destroyed, so when the call is finally established between A and C there are two active dialogs as I could check with `kamctl fifo get_statistics active_dialogs`. Shouldn't there be only one active dialog? Is this a problem in my script or a problem with the dialog module? Thanks in advance, Santi

3 3

[Serusers] tell me module of ser for nideo streaming..
by city spider 18 Feb '10

18 Feb '10

i want to use ser for streaming . please tell me the specific module for streaming... in streaming me using RTSP (Real Time Streaming Protocol) suggest me the specific module that will be added to fulfil my purpose.

1 0

[Kamailio-Users] Help me on 'No matching subscription dialog found'
by Asterisk User 18 Feb '10

18 Feb '10

Hi Group, I have one more problem regarding a SIP subscribe packet. When my phone registers first time, it gets OK for SUBSCRIBE sent from phone. But after couple of hours it(phone) keeps on sending subsequent SUBSCRIBE requests but the response from Kamailio is 481 Subscription does not exist. Here are my packets, =======================> For first SUBSCRIBE SUBSCRIBE sip:ABC.com SIP/2.0 Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-4082dbb From: DEF <sip:307@ABC.com>;tag=802bf23e23f4741 To: DEF <sip:307@ABC.com> Call-ID: 9f465e4b-2207f239(a)172.18.100.89 CSeq: 30004 SUBSCRIBE Max-Forwards: 70 Contact: DEF <sip:307@172.18.100.89:5060> Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 ------------------------------------------------------- SIP/2.0 202 OK Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-4082dbb;rport=5060;received=220.224.237.54 From: DEF <sip:307@ABC.com>;tag=802bf23e23f4741 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 9f465e4b-2207f239(a)172.18.100.89 CSeq: 30004 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.94.30:5060> Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 =======================> For subsequent SUBSCRIBEs SUBSCRIBE sip:ABC.com SIP/2.0 Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-69712d2e From: DEF <sip:307@ABC.com>;tag=5d3bd6c2989f0839 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 7f6d2c36-97569155(a)172.18.100.89 CSeq: 55947 SUBSCRIBE Max-Forwards: 70 Contact: DEF <sip:307@172.18.100.89:5060> Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 ------------------------------------------------------- SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-69712d2e;rport=5060;received=220.224.237.54 From: DEF <sip:307@ABC.com>;tag=5d3bd6c2989f0839 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 7f6d2c36-97569155(a)172.18.100.89 CSeq: 55947 SUBSCRIBE Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 ---------------------------------------------------------------------------------------------------------------------------------------------- ===>The error which I get on kamailio is, ERROR:presence:get_database_info: No matching subscription dialog found in database ===>Have already loaded the module presence.so modparam("presence", "db_url", "mysql://XYZ:XYZ@192.168.90.12/openser") modparam("presence", "max_expires", 3600) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "db_update_period", 100) modparam("presence", "to_tag_pref", 'pres') modparam("presence", "expires_offset", 10) modparam("presence", "server_address", "sip:192.168.94.30:5060") modparam("presence", "fallback2db", 1) modparam("presence", "subs_htable_size", 11) modparam("presence", "pres_htable_size", 11) loadmodule "presence_mwi.so" loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) modparam("presence_xml", "db_url", "mysql://XYZ:XYZ@192.168.90.12/openser") --------------------------------------------------------------------------------------------------------------------------------------------- Please help me on this. --SM

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[Kamailio-Users] starting
by Andrew Matthews 17 Feb '10

17 Feb '10

I'm looking for a tutorial of some sort to help me with getting openser running. What I need from it, are a basic proxy with both IP and username/password authentication, and an lcr. I've searched the net and can't seem to find what I need. I'm new to openser, I came from the asterisk world. If anyone can point me in the right direction I'd appreciate it. Thanks, Exstatica

2 1

Number Portability Functionality
by Asim Riaz 17 Feb '10

17 Feb '10

Hi list, Can someone explain me about PDB server? what I understand that PDB server have all the routing information for all the numbers and SIP-Router receive that information and route the call accordingly but how PDB server receive the routing data? Do we need to load this routing data manually? What happen if the routing information for a number is changed, do we need to update manually? Thanks, Asim

3 2

[Kamailio-Users] Inbound call no way audio
by RAJNIKANT VANZA 17 Feb '10

17 Feb '10

Hi Friends, Outbound calling works fine. When I receive an inbound call, the phone rings, but when I pick up the phone, the conversation does not get established. I have call from inside NAT extension number to Outside NAT extension ( Outboud call ) then call get established and two-way audio work fine but when I have call from Outside NAT extension to Inside NAT extension (Inbound call ) then phone rings, but when pickup the phone, audio is not present. During inboud call RTP packets are not transmit whereas Outboud call RTP packets transmit successful. Sometimes, Inbound and Outbound call only one-way audio present. Can anybody help me, or point me to a previous thread where this issue was discussed and solved? Thanks in advance. -- Best Regards, Rajnikant Vanza

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[Kamailio-Users] What is the best (and secure) way to change FROM_URI using regexp?
by Fernando Frota M Morais 16 Feb '10

16 Feb '10

Before sending this message I looked for an answer and found some options, two of them are followed: 1) SUBST function, sample listed here: if (is_present_hf("User-Agent")) { remove_hf("User-Agent"); remove_hf("P-Asserted-Identity"); subst("/^From: \".+\" (<sip: *[a-zA-Z0-9_\.-]+ <at> .*>)/From: \1/ig"); subst("/^To: \".+\" (<sip: *[a-zA-Z0-9_\.-]+ <at> .*>)/To: \1/ig"); } 2-a) uac_replace: if (from_uri=~"^sip:10804007\@voip.mydomain") { xlog("L_ALERT", "TESTANDO - Original From is ($fu-$fn)\n"); # uac_replace_from("sip:031" . $fU . "\(a)voip.mydomain"); uac_replace_from("Frota10804007","sip:03110804007@voip.mydomain"); xlog("L_ALERT", "TESTANDO - From changed to ($fu-$fn)\n"); }; OR : 2-b) uac _replace #2: if (from_uri=~"^sip:10804007\@.*") { xlog("L_ALERT", "Route 21 - Original from is ($fu-$fn)\n"); $avp(from)=$fu; # $avp(to)=$ru; # avp_pushto("$ru", "$avp(from)"); avp_subst("$avp(from)", "/(sip:)/sip:031/"); uac_replace_from("$avp(from)"); # xlog("L_ALERT", "Route 21 - From changed to ($fu-$fn)\n"); # avp_pushto("$ru", "$avp(to)"); }; } OR Looks like uac_replace is safer then "subst" function, because it can be reverted, but I did not found a way to use concatenation or any other kind of string operation as a parameter for it. I don't want to start a discussion, I need only a few lines showing me how to change from_uri_from from "sip:108xxxx" to "sip:031108xxxx". Thanks in advance, -- Fernando Frota Machado de Morais Divisao de Redes de Comunicacao Centro de Computacao Universidade Federal de Minas Gerais Brasil Tel. +55(31)3409.4007 Fax. +55(31)3409.4004

2 1

[Kamailio-Users] What is wrong with this NOTIFY packet
by Asterisk User 15 Feb '10

15 Feb '10

Hi Group, I know this issue has to do bit more with SIP and off from kamailio problems, but just wanted to know your opinions on this. I have my sip extensions registered to kamilio and Asterisk works as a voicemail server. Now to implement MWI i send NOTIFY packet from Asterisk to Kamailio and from kamailio to SIP extension (Grandstream GXP2000). here is the packet that travels from kamailio to extension and the response sent from extension to kamilio. =========================== NOTIFY sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com> SIP/2.0 Max-Forwards: 10 Record-Route: <sip:172.18.100.74;lr;nat=yes> Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bK3c3.e1003f03.0 Via: SIP/2.0/UDP 127.0.0.1:45382 ;received=172.18.100.73;branch=z9hG4bK.4f83530d;rport=45382;alias Via: SIP/2.0/UDP 172.18.100.73:5060 From: <sip:111@arzoo.com <sip%3A111(a)arzoo.com>> To: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>> Contact: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>> Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)172.19.100.42 CSeq: 1 NOTIFY User-Agent: VoiceMail Event: message-summary Content-Type: application/simple-message-summary Content-Length: 39 =========================== SIP/2.0 415 Unacceptable Content-Type Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bK3c3.02003f03.0 Via: SIP/2.0/UDP 127.0.0.1:43416 ;received=172.18.100.73;branch=z9hG4bK.7be73901;rport=43416;alias Via: SIP/2.0/UDP 172.18.100.73:5060 Record-Route: <sip:172.18.100.74;lr;nat=yes> From: <sip:111@arzoo.com <sip%3A111(a)arzoo.com>> To: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>>;tag=3c6ba3e4a93d6864 Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)172.19.100.42 CSeq: 1 NOTIFY User-Agent: Grandstream GXP2000 1.1.4.14 Accept: application/sdp, application/simple-message-summary, application/octet-stream, application/dialog-info+xml, application/pidf+xml, message/sipfrag;version=2.0 Content-Length: 0 =========================== Please guide me in right direction to get rid of 'Unacceptable Content-Type' problem. Thanks in advance, --SM

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[Kamailio-Users] Problems with parallel forking to different users
by Anton Roman 15 Feb '10

15 Feb '10

Hi all, I'm trying to implement parallel forking to different users registered on a Kamailio server. I want to fork the call to a number of different users (not including the called user). To do this I added the piece of code below to the default 3.0.0 config file: ... # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting setflag(4); #dialog aware if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); }; } .... This is usr_preference table content: +----+------+----------+--------+-----------+------+-------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+------+----------+--------+-----------+------+-------+---------------------+ | 4 | | 1000 | | fork | 0 | 1001 | | | 3 | | 1000 | | fork | 0 | 1005 | | +----+------+----------+--------+-----------+------+-------+---------------------+ Kamailio correctly forks the call to 1001 and 1005, but the Request-URI of the 1005 user is not correctly set, because the proxy IP is used instead of the IP where the user is registered (10.1.2.15). You can see the INVITEs below. Could anybody tell me what I'm doing badly? Thanks for your time, best regards Anton ---------------------------------------------------------------------------------------------------------------------------------------- # U +0.000328 10.1.20.103:5060 -> 10.1.3.18:5061 INVITE sip:1001@10.1.3.18:5061 SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=c34.3ed81d77> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKc40b.ec3572a3.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-213aa77 From: "1002" <sip:1002@10.1.20.103 <sip%3A1002(a)10.1.20.103> >;tag=280da520d15d62cfo1 To: "1000" <sip:1000@10.1.20.103 <sip%3A1000(a)10.1.20.103>> Call-ID: 8973c34-427096cb(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "1002" <sip:1002@10.1.3.15:5061> Expires: 240 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 202 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 2003696 2003696 IN IP4 10.1.3.15 ... # U +0.000052 10.1.20.103:5060 -> 10.1.20.103:5060 INVITE sip:1005@10.1.20.103 <sip%3A1005(a)10.1.20.103> SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=c34.3ed81d77> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKc40b.ec3572a3.1 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-213aa77 From: "1002" <sip:1002@10.1.20.103 <sip%3A1002(a)10.1.20.103> >;tag=280da520d15d62cfo1 To: "1000" <sip:1000@10.1.20.103 <sip%3A1000(a)10.1.20.103>> Call-ID: 8973c34-427096cb(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "1002" <sip:1002@10.1.3.15:5061> Expires: 240 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 202 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 2003696 2003696 IN IP4 10.1.3.15 ... ----------------------------------------------------------------------------------------------------------------------------------------

2 1

[Kamailio-Users] Registration Timeout Retransmission
by Hemanshu Patel 13 Feb '10

13 Feb '10

Hello to all users, I am very new to kamailio and just started using it. I have installed it and my phones are getting register successfully. But sometimes i am seeing a scenario, which makes my phones unusable for sometime. please see the condition as below | phone |--------------| Kamailio | | REGISTER-----------------> <----------------------401 | REGISTER-----------------> Tricky condition here | timeout <--------------200 OK phone shows itself as unregistered | REGISTER----------------> <--------------------401 After several retries phone stops to retry till say next 20 seconds, during that time though there's an entry in location table in kamailio (i am using mysql) my phone shows itself as unregistered phone. Due to load or any other conditions Server takes a bit time to send 200OK back to phone and During that time timer expires and phone sends REGISTER once again. now phone will keep sending REGISTER request with same nonse and other value, and kamailio is configured to not to entertained same credentials with nonce if it is authenticated once. So kamailio generates new 401 Message with new nonce value to which phone understands as the previous reply thet he send for previous 401 was rejected, and the most probable reason is wrong user credentials..... Hope you guys have got my problem? i can not allow kamailio to let use the same credentials+ response with nonce if it has used it once for security reason. Then what could be the solution for this. I am very good in Programming, but just that i have just started to use kamailio and not very much aware about its architecture. -- Regards, Hemanshu Patel M: 09601295238

3 5

[Serusers] stateful statistics
by Sabatella, Michael 12 Feb '10

12 Feb '10

I know somebody else posted this a while ago but I did not see any response. I tried to research it but could not come up with any answers and would like to get an understanding of this . On the stateful statistics what does 130 waiting and 6 current mean Stateful Statistics Current: 6 (130 waiting) Total: 3540496 (0 local) Replied localy: 3809867 Is it recommended to restart the process once a week or once a month? I am running ser 0.9.6 Thanks for the help. Michael ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

1 0

[Kamailio-Users] Error in header - where and how?
by Anders 12 Feb '10

12 Feb '10

Hi, My carrier doesn't like the invites that come from two specific customers I have, who both use something called VoipSwitch. It returns from the carrier as a Bad Request. Below is the dialog - I'm not sharp enough to see what's wrong - anyone? Thanks! //Anders U 2010/02/01 17:53:42.355689 CustomerIP:5060 -> MyProxyIP:5060 INVITE sip:890912551128650136@MyProxyIP SIP/2.0. CSeq: 1 INVITE. Via: SIP/2.0/UDP CustomerIP:5060;branch=z9hG4bk01025810150801381470578. From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Contact: <sip:CustomerIP:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 250. Remote-Party-ID: "SERVER BR"<sip:69912390@customerDNS>;privacy=off;screen=no. . v=0. o=VoipSwitch 9298 9298 IN IP4 CustomerIP. s=VoipSIP. i=Audio Session. c=IN IP4 CustomerIP. t=0 0. m=audio 8298 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2010/02/01 17:53:42.355990 MyProxyIP:5060 -> CustomerIP:5060 SIP/2.0 100 Giving a try. CSeq: 1 INVITE. Via: SIP/2.0/UDP CustomerIP:5060;branch=z9hG4bk01025810150801381470578;rport=5060;received=187.45.218.17$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Server: Kamailio. Content-Length: 0. . U 2010/02/01 17:53:42.356010 MyProxyIP:5060 -> CarrierIP:5060 INVITE sip:84370551128650136@CarrierIP:5060 SIP/2.0. Max-Forwards: 10. Max-Forwards: 5. Record-Route: <sip:MyProxyIP;lr=on;nat=yes;did=e4c.e1f33ad5>. CSeq: 1 INVITE. Via: SIP/2.0/UDP MyProxyIP;branch=z9hG4bK139a.a74691c5.0. Via: SIP/2.0/UDP CustomerIP:5060;rport=5060;received=CustomerIP;branch=z9hG4bk0102581015080138147057$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Contact: <sip:CustomerIP:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 250. Remote-Party-ID: "SERVER BR"<sip:69912390@customerDNS>;privacy=off;screen=no. . v=0. o=VoipSwitch 9298 9298 IN IP4 CustomerIP. s=VoipSIP. i=Audio Session. c=IN IP4 CustomerIP. t=0 0. m=audio 8298 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2010/02/01 17:53:42.608917 CarrierIP:5060 -> MyProxyIP:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP MyProxyIP;branch=z9hG4bK139a.a74691c5.0. Via: SIP/2.0/UDP CustomerIP:5060;rport=5060;received=CustomerIP;branch=z9hG4bk0102581015080138147057$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. To: <sip:890912551128650136@MyProxyIP>. Call-ID: GW6d4c945d2c573c27@CustomerIP. CSeq: 1 INVITE. Content-Length: 0.

2 4

[Kamailio-Users] Drouting Question
by Sven Schulz 12 Feb '10

12 Feb '10

Hello, Ive been testing Kamailio 3.0 and drouting. In the documentation it states (Section 1.5 Routing Rule Processing) ³The module will execute serial forking for each address in chain. The next address in chain is used only if the previously has failed.² What exactly defines a failure? Does it use ping or OPTIONS to determine if a GW is busy or down? Id like to know exactly how drouting figures this out. Sven Schulz Penn State University Telecommunications and Network Services 814.865.6116 sip:sven@psu.edu

3 4

[Kamailio-Users] acc module
by Denis Putyato 12 Feb '10

12 Feb '10

Hello everybody! I have a such problem with ACC module. Calls was accounted but only INVITE and BYE requests has appeared in acc table. ACK is absent. Kamailio 3.0 was installed from GIT. Thank you for any help loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "siptrace.so" loadmodule "dialplan.so" loadmodule "lcr.so" loadmodule "sqlops.so" loadmodule "carrierroute.so" loadmodule "drouting.so" loadmodule "kex.so" # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 29) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_url", "xxxxxxxxxxxxxxxxxxxxxxxx") modparam("acc", "db_extra", "src_user=$avp(i:2);src_domain=$si;dst_ouser=$avp(i:1);dst_user=$rU;dst_doma in=$rd") route { sip_trace(); setflag(30); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(29); } if (is_method("ACK")) { xlog("L_NOTICE", "ACK detected"); # I see this line in syslog setflag(1); setflag(29); } route("relay"); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { setflag(1); route("relay"); exit; } else { exit; } } } }

1 0

[Kamailio-Users] acc & radius
by Kris Amy 12 Feb '10

12 Feb '10

Hi all, Is there anything special required to get radius support into the acc module? I have tried the following... make distclean make config modules_include="mediaproxy auth_radius group_radius uri_radius misc_radius" group_include="mysql standard" make deb But alas it still seems to not have it. I have confirmed that the radius libraries are installed aswell. ii libradiusclient-ng-dev 0.5.6-1 Enhanced RADIUS client library development f ii libradiusclient-ng2 0.5.6-1 Enhanced RADIUS client library Cheers, Kris

2 1

[Kamailio-Users] Extra byte in Content-length header
by Vikram Ragukumar 11 Feb '10

11 Feb '10

Hello, I am testing a cellphone software that registers with an Asterisk server through an outbound proxy (Kamailio+rtpproxy). I am capturing network traffic using wireshark and i notice that the SIP Register message from the cellphone has an extra byte in the Content-Length header, which causes Kamailio to discard the message with the following error output : - ERROR:core:parse_content_length: parse error near char [13][^M] - ERROR:core:get_hdr_field: bad content_length header Could it be that this byte was appended somewhere downstream(possibly at the wireless router..) or can i be sure that the byte was added during SIP Message construction by the cellphone software ? Thanks in advance, Vikram.

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Re: [Kamailio-Users] Siremis Accounting
by Daniel-Constantin Mierla 11 Feb '10

11 Feb '10

Hello, please use public mailing list for such questions. Direct messages get long delayed or lost in most of the cases. Mailing list has plenty of developers and community users that can help you faster. Besides that, new people can benefit of older discussions. On 2/11/10 2:14 AM, Kris Amy wrote: > Hi Daniel, > > It would seem to me that the stored procedure on > http://siremis.asipto.com/install-accounting/ is for PostgreSQL. I'm > going to try and getting it working with MySQL. it is for mysql. Cheers, Daniel > > Just thougth you may want to update the page. > > -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 * http://www.asipto.com/

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Re: [Kamailio-Users] Siptrace and Kamailio 3.0.0
by Daniel-Constantin Mierla 11 Feb '10

11 Feb '10

On 2/11/10 11:16 AM, Kris Amy wrote: > On Thu, Feb 11, 2010 at 8:09 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > ok, then the only point in mod_init of siptrace that causes > initialization error without printing a log message is when you > haven't set the trace_flag parameter: > http://kamailio.org/docs/modules/3.0.x/modules_k/siptrace.html#id2520001 > > Cheers, > Daniel > > > Hi Daniel, > > That fixed it. > > Perhaps there is a bug in that it's not picking up the default of 0? in the sources the default is -1, which means disabled -- docs needs to be updated. Thanks for pointing, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 *http://www.asipto.com/

1 0

[Kamailio-Users] SDP IP address not rewritten
by Vikram Ragukumar 11 Feb '10

11 Feb '10

Hello, I am running Kamailio version 1.5 and rtpproxy version 1.2. I have enabled NAT support in Kamailio.cfg. I have a UA behind NAT (UA IP:192.168.2.101), and the UA registers with an Asterisk server (Public IP) through an outbound proxy running Kamailio and rtpproxy with Public IP. Registration succeeds, however when making an outbound call from the UA to a cellphone, there is only one sided voice. This is happening because Asterisk is addressing rtp streams to the UA's local IP address instead of its PublicIP:port. Why is the IP address not being rewritten in the SDP ? Is there an alternate way of invoking rtpproxy (with flags) such that the IP address in the SDP is rewritten ? Thanks and Regards, Vikram. PS : Below is the main routing section of my kamailio.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (isflagset(5)) { setbflag(6); # setbflag(7); } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); setflag(6);#VR } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } }

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[Kamailio-Users] [kamailio-users] Method NOTIFY not supported ...
by Chandrakant Solanki 11 Feb '10

11 Feb '10

Hi below line is in my configuration... loadmodule "pua.so" modparam("pua", "db_url", "mysql:// kamailio:kamailio@172.16.0.10:3306/kamailio") modparam("pua", "db_table", "pua") ---- ---- if(is_method("NOTIFY") && src_ip==172.16.0.15) { xlog("L_INFO", "NOTIFY Method found\n"); if(!t_relay()){ sl_reply_error(); } } ---- ---- I found following error in log... Feb 10 15:14:40 kamailio_1 /sbin/kamailio[5597]: Method not supported - M=NOTIFY RURI=sip:172.16.0.15 F=sip:2222222@pbxusers.net<sip%3A2222222(a)pbxusers.net>T=sip:172.16.0.15 IP=172.18.100.89 ID= 17dd4bb9-81d132a0(a)172.18.100.89 -- Regards, Chandrakant Solanki

2 7

[Kamailio-Users] Fwd: Siptrace and Kamailio 3.0.0
by Kris Amy 11 Feb '10

11 Feb '10

---------- Forwarded message ---------- From: Kris Amy <kris(a)amy.id.au> Date: 2010/2/11 Subject: Re: [Kamailio-Users] Siptrace and Kamailio 3.0.0 To: Daniel-Constantin Mierla <miconda(a)gmail.com> On Thu, Feb 11, 2010 at 8:09 PM, Daniel-Constantin Mierla <miconda(a)gmail.com > wrote: > Hello, > > ok, then the only point in mod_init of siptrace that causes initialization > error without printing a log message is when you haven't set the trace_flag > parameter: > http://kamailio.org/docs/modules/3.0.x/modules_k/siptrace.html#id2520001 > > Cheers, > Daniel >

1 0

[Kamailio-Users] Siptrace and Kamailio 3.0.0
by Kris Amy 11 Feb '10

11 Feb '10

Hi, Just wondering if anyone is using Siptrace with Kamailio 3.0.0. The module fails to load and I have no idea why (or how to get more information than from debug = 99). 0(13453) DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: siptrace 0(13453) ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module siptrace ERROR: error while initializing modules Cheers, Kris

2 4

[Kamailio-Users] kamailio crashes during connection attempt
by Dmitri 10 Feb '10

10 Feb '10

Hello list, I have fresh installation of Debian Squeeze on X86_64. I downloaded source tarball http://www.kamailio.org/pub/kamailio/latest/src/kamailio-3.0.0_src.tar.gz and did 'make deb' there. So, I got following debian packages: kamailio_3.0.0_amd64.deb, kamailio-berkeley-modules_3.0.0_amd64.deb, kamailio-carrierroute-modules_3.0.0_amd64.deb kamailio-cpl-modules_3.0.0_amd64.deb, kamailio-ldap-modules_3.0.0_amd64.deb, kamailio-memcached-modules_3.0.0_amd64.deb kamailio-mysql-modules_3.0.0_amd64.deb, kamailio-nth_3.0.0_amd64.deb, kamailio-perl-modules_3.0.0_amd64.deb kamailio-postgres-modules_3.0.0_amd64.deb, kamailio-presence-modules_3.0.0_amd64.deb, kamailio-purple-modules_3.0.0_amd64.deb kamailio-radius-modules_3.0.0_amd64.deb, kamailio-snmpstats-modules_3.0.0_amd64.deb, kamailio-tls-modules_3.0.0_amd64.deb kamailio-unixodbc-modules_3.0.0_amd64.deb, kamailio-utils-modules_3.0.0_amd64.deb, kamailio-xmlrpc-modules_3.0.0_amd64.deb kamailio-xmpp-modules_3.0.0_amd64.deb I installed packages: kamailio_3.0.0_amd64.deb, kamailio-mysql-modules_3.0.0_amd64.deb, kamailio-presence-modules_3.0.0_amd64.deb, kamailio-xmlrpc-modules_3.0.0_amd64.deb Created database, changed passwords for openser and openserro users in MySQL as well as in kamailio config files, enabled MySQL. DUMP_CORE=yes in /etc/default/kamailio, also added '-w /var/run/kamailio' to options. Created test user with kamctl. Started kamailio with init script: /etc/init.d/kamailio start: Feb 7 22:05:10 kamailio kamailio: INFO: <core> [tcp_main.c:4150]: init_tcp: using epoll_lt as the io watch method (auto detected) Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: usrloc [hslot.c:53]: locks array size 512 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2695]: INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config) Then I tried to connect to kamailio server with softphone Zoiper, and kamailio crashes immediately: Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2700]: : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 21 Feb 7 22:05:20 kamailio kernel: [ 1051.407643] kamailio[2689]: segfault at c2 ip 00007fa8b5947b19 sp 00007fff48ff9700 error 4 in tm.so[7fa8b5917000+72000] Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: ALERT: <core> [main.c:722]: child process 2689 exited by a signal 11 Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: ALERT: <core> [main.c:725]: core was not generated Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2691]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2699]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2698]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2687]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2697]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2690]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2696]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2688]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2695]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2680]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2694]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2693]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2692]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2686]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2685]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2681]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2684]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2682]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2700]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2683]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted (1) Feb 7 22:05:20 kamailio kernel: [ 1051.432646] kamailio[2678]: segfault at 12 ip 00007fa8b592ae43 sp 00007fff48ff9f40 error 4 in tm.so[7fa8b5917000+72000] For unknown reason core was not generated. Any ideas ? Many thanks in advance, Dmitri

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[Kamailio-Users] save() function
by Kris Amy 10 Feb '10

10 Feb '10

Hi All, It was to my understanding that this function was part of the core and thus didn't come from a module. The relevant code snippet is if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } The error I'm getting is.. 0(20467) DEBUG: <core> [sr_module.c:516]: find_export_record: <save> not found 0(20467) DEBUG: <core> [sr_module.c:516]: find_export_record: <save> not found 0(20467) : <core> [cfg.y:3329]: parse error in config file /etc/kamailio/kamailio.cfg, line 656, column 21: unknown command, missing loadmodule? Cheers, Kris

2 2

[Kamailio-Users] Carrierroute issue
by michel freiha 10 Feb '10

10 Feb '10

Hi all, I'm using carrierroute module and ie seems that it's not working well...Suppose I have 2 entries in carrierroute table...Entry 1 with scan prefix 00 and prob 0 and a second entry with scan prefix 00 and prob=1...As soon as i make a call, this call will be forwarded to the entry with prob 0 because i guess it appears before the one with prob 1 in the table... When starting Kamailio I got the below: Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 1, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: domain 0 not found, add it Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_tree: tree default has 1 trees Jul 29 12:30:11 [23143] DBG:carrierroute:add_route_tree: tree 0Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: created route tree: 0, with id 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 00, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 111, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 00, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules It seems that kamailio is not reading probability entry in carrier route table Please find below carrierroute table structure: CREATE TABLE IF NOT EXISTS `carrierroute` ( `id` int(10) unsigned NOT NULL auto_increment, `carrier` int(10) unsigned NOT NULL default '0', `domain` varchar(64) NOT NULL default '', `scan_prefix` varchar(64) NOT NULL default '', `flags` int(11) unsigned NOT NULL default '0', `mask` int(11) unsigned NOT NULL default '0', `prob` float NOT NULL default '0', `strip` int(11) unsigned NOT NULL default '0', `rewrite_host` varchar(128) NOT NULL default '', `rewrite_prefix` varchar(64) NOT NULL default '', `rewrite_suffix` varchar(64) NOT NULL default '', `description` varchar(255) default NULL, PRIMARY KEY (`id`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=23 ; Can someone help me to find this issue? Regards

5 13

[Kamailio-Users] check_user_blacklist function
by Antonio Reale 10 Feb '10

10 Feb '10

Hi all, the function check_user_blacklist in kamailio 1.4 returns -1 (FALSE based on the doc) "if a match is found and it is not set to whitelist", otherwise it returns 1 (TRUE). So the function returns the same value if a match is found and is set to whitelist and if there are no matches. I need to differentiate the two situations. The userblacklist module in kamailio 3.0 has a new function "check_user_whitelist" that can solve my problem, but at this time I can't upgrade. I solved the problem, changing only the return value in case of whitelisted entry (userblacklist.c file), but I don't know if it is a safe (and suffice) modification. Below the piece of code in the userblacklist.c: if (dt_longest_match(dt_root, ptr, &whitelist) >= 0) { if (whitelist) { /* LM_ERR("whitelisted"); */ return 2; /* found, but is whitelisted */ <===== CHANGED HERE. IT WAS return 1 } } else { /* LM_ERR("not found"); */ return 1; /* not found is ok */ } Thanks. Regards. -- Antonio

2 2

[Kamailio-Users] LCR
by Kris Amy 10 Feb '10

10 Feb '10

Hi All, I've been ripping my hair out for a little with load_gws(). How do I know what the lcr_id is and the caller_uri? I would have thought that $fu for caller_uri would have been sufficient but it seems not. Cheers, Kris

3 2

[Kamailio-Users] kamctl and lcr table
by Kris Amy 10 Feb '10

10 Feb '10

Hi, Could we get the lcr-create.sql updated to have the ping field? Cheers, Kris

2 3

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 09 Feb '10

09 Feb '10

Hello, I was thinking of having an IRC meeting to discuss next steps within sip router. Now that 3.0 branches are out for a while, would be good to plan a bit 3.1. For me Friday (or even Thursday if not too early for others to make arrangements) this week would be good, say 14:00 or 15:00UTC to allow fair time for US. If this date does not fit, please make proposals. The channel is #sip-router on freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 * http://www.asipto.com/

1 0

[Kamailio-Users] problems with mi_xmlrpc command
by Anton Roman 09 Feb '10

09 Feb '10

Hi all, I'm trying to send xmlrpc commands to Kamailio with Siremis but I get the error below: Fault Code: 5 Fault Reason: Didn't receive 200 OK from remote server. (HTTP/1.1 400 Bad Request) I get the same error when I manually send the xmlrpc command with curl. On the other hand, mi_xmlrpc module seems to be correctly loaded, but nothing is written in the log file (set with modparam("mi_xmlrpc", "log_file", "/var/log/abyss.log")). Does anybody know where the problem might be? Best regards, thanks in advance. Anton Roman These are the traces of the xmlrpc message flow: T +11.569284 127.0.0.1:37821 -> 127.0.0.1:8080 [AP] POST HTTP/1.0 User-Agent: PEAR XML_RPC Host: 127.0.0.1 Content-Type: text/xml Content-Length: 135 <?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>system.listMethods</methodName> <params> </params> </methodCall> ## T +0.000363 127.0.0.1:8080 -> 127.0.0.1:37821 [AP] HTTP/1.1 400 Bad Request # T +0.000146 127.0.0.1:8080 -> 127.0.0.1:37821 [AP] Content-type: text/html # T +0.001982 127.0.0.1:8080 -> 127.0.0.1:37821 [AFP] Connection: close Date: Mon, 08 Feb 2010 14:04:08 GMT Server: XMLRPC_ABYSS/1.06 <HTML><HEAD><TITLE>Error 400</TITLE></HEAD><BODY><H1>Error 400</H1>Bad Request<HR><a href="http://xmlrpc-c.sourceforge.net">ABYSS Web Server for XML-RPC For C/C++</a> version 1.06 </BODY></HTML> ####

2 2

[Kamailio-Users] Stateless forward
by Antonio Reale 09 Feb '10

09 Feb '10

Hi all, I'm trying to setup a very basic scenario with kamailio 3.0. It is placed in front of a softswitch and it has to modify some headers before forward (stateless) messages to softswitch. If I use route { rewritehost("192.168.10.1"); t_relay(); } all work, but if I want to do that stateless and I use forward() or send() instead of t_relay(), I get a syntax error: "bad forward argument" or "bad send argument". If I use ... forward("udp:192.168.10.1:5060"); ... I see in the log /usr/local/sbin/kamailio[3675]: : <core> [proxy.c:271]: ERROR: mk_proxy: could not resolve hostname: "udp:192.168.10.1:5060" /usr/local/sbin/kamailio[3675]: ERROR: <core> [route.c:1026]: fixing failed (code=-478) at cfg:/usr/local/etc/kamailio/kamailio.cfg:77 Same error using the long format of send() function. I can't see any particular hint on the documentation, what I'm missing? Thank you. Best regards. -- Antonio

3 6

[Kamailio-Users] Install Documentation
by Kris Amy 09 Feb '10

09 Feb '10

Hi All, Just thought that you might want to put down that libpcre is required for lcr in the install document. (Since the other main modules have their requirements listed). Cheers, Kris

2 1

[Kamailio-Users] change domain at to header
by Adnan Çelikel 09 Feb '10

09 Feb '10

Dear All, I want to change a field (Sip to address host part) at sip message header. I think avpops will solve this.If anybody has a sample confg and send to me I will be very glad.Regards,Adnan. #---------------------------------------------------- #7777 CALL ROUTING #-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-* if(is_method("INVITE") && (uri=~"sip:7777")){ rewritehostport("192.168.1.10:5060"); route(1); exit; } #---------------------------------------------------- at this rule, the invite is going to 192.168.1.10 with the host part as abc.com, we want to send with another domain. Instead of abc.com domain, I want to use 192.168.1.10 domain ; To:<sip:7777@abc.com> To:<sip:7777@192.168.1.10> _________________________________________________________________ Windows Live Hotmail: Arkadaşlarınız Facebook'taki güncellemelerinizi doğrudan Hotmail®'den alır. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-networ…

5 11

Routing Problems
by Andreas Rehbein 08 Feb '10

08 Feb '10

Hello, we use Kamailio 3.0 on a Red Hat EL5.4 system with openssl 0.9.8e (the current Red Hat OpenSSL version). We want to use Kamailio 3.0 in front of our Call Server (OpenSER 1.3.2) as a security gateway. So the Call Server should only deal tcp, while the Security Gateway terminates the TLS Data which he receives from the User Agents but forwards the SIP-Messages via TCP to the Call Server. UA ---SIP_over_TLS---> Security Gateway (Kam3.0) ---SIP_over_TCP---> Call Server (OpenSER1.3.2) UA<---SIP_over_TLS--- Security Gateway (Kam3.0) <---SIP_over_TCP--- Call Server (OpenSER1.3.2) Nearly everything works fine in this scenario: User Agents are able to register and when they are sending INVITES the callee receives it. The problem we have right now is: if Kamailio 3.0 receives BYE we get 477 Unfortunatly error on sending to next hop occured. It seems that the messages are ok... I attached a text file with the BYE message and the errors. Explanation for the text file: * UA1: 192.168.0.126 * UA2: 192.168.0.176 *Security Gateway (Kam3.0): 192.168.0.89 *Call Server (OpenSER1.3.2): 192.168.0.106 Do you have any suggestions? Thank you very much in advance! Regards Andreas Dipl.-Ing. (FH) Andreas Rehbein FH Frankfurt am Main - University of Applied Sciences FB 2 - Informatik und Ingenieurwissenschaften Forschungsgruppe für Telekommunikationsnetze Kleiststraße 3 60318 Frankfurt am Main Tel.: +49 (0) 69 1533 3613 Fax: +49 (0) 69 1539 5224 961 E-Mail: <mailto:rehbein@e-technik.org> rehbein(a)e-technik.org Internet: <http://www.e-technik.org/frame_forschung.htm> http://www.e-technik.org/frame_forschung.htm

2 2

[Kamailio-Users] Maximum retries exceeded on transmission
by Daniel-Constantin Mierla 06 Feb '10

06 Feb '10

On Sat, Feb 6, 2010 at 11:10 AM, Jon Farmer <viperdudeuk(a)gmail.com> wrote: > Sorry it was a problem I had with gmail. > It happens to me from time to time, but I couldn't find the real cause, whether is gmail or thunderbird. Thunderbird hangs a while, reports a delivery error, then message appears twice... Daniel > On 5 Feb 2010 12:21, "Iñaki Baz Castillo" <ibc(a)aliax.net> wrote: > > 2010/2/5 Jon Farmer <viperdudeuk(a)gmail.com>: > > Hi > > Please don't send the same smail twice. Thanks. > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Maximum retries exceeded on transmission
by Daniel-Constantin Mierla 06 Feb '10

06 Feb '10

Hello, On Fri, Feb 5, 2010 at 10:56 AM, Jon Farmer <viperdudeuk(a)gmail.com> wrote: > Hi > > I am getting this dreaded error on a asterisk box when call from one > UA to another using Asterisk as a media relay and openser as a SBC and > proxy. > > chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission > 1310650592(a)10.100.0.101 > > If I make a call from a UA to the PSTN through Asterisk all is > fine. This only happens when calling from one UA to another on the > same LAN. > > I have posted an example sip trace here http://pastebin.com/m2262b0ef > ACK is not accepted as I could spot quickly, 200ok being retransmitted. Probably there is something wrong with dialog attributes, after Fosdem I can check it better. Daniel > My openser.cfg is here http://pastebin.com/d5e84aaf7 > > I realise this is a issue with Asterisk but I have not been able to > find a fix. Therefore I figure there might be something I can do on > openser. The problem seems to be with a ACK not being returned but my > SIP knowledge is not good enough to know how to fix it. Any help is > greatly appreciated. > > Regards > > Jon > -- > Jon Farmer > Tel: 07795 118140 > Email: viperdudeuk(a)gmail.com > Twitter: @viperdudeuk > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Maximum retries exceeded on transmission
by Jon Farmer 06 Feb '10

06 Feb '10

Hi I am getting this dreaded error on a asterisk box when call from one UA to another using Asterisk as a media relay and openser as a SBC and proxy. chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1310650592(a)10.100.0.101 If I make a call from a UA to the PSTN through Asterisk all is fine. This only happens when calling from one UA to another on the same LAN. I have posted an example sip trace here http://pastebin.com/m2262b0ef My openser.cfg is here http://pastebin.com/d5e84aaf7 I realise this is a issue with Asterisk but I have not been able to find a fix. Therefore I figure there might be something I can do on openser. The problem seems to be with a ACK not being returned but my SIP knowledge is not good enough to know how to fix it. Any help is greatly appreciated. Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

2 2

[Serusers] RECEIVED column in SER.LOCATION table
by Bill McNamara 05 Feb '10

05 Feb '10

Question, when does a value get inserted into the ser.location table’s received column. We have some UA’s with a contact like value and some with null values – different devices (Polycom 331/450/55, Grandstream 286/486) behind the same firewall. I’m just curious to know why some have a received value and some don’t. Thanks, Bill SER 2.0 Linux 2.6.9-78 x86_64 MySQL 5.1.30

2 1

[Kamailio-Users] CRITICAL:dialog:log_next_state_dlg
by alex pappas 05 Feb '10

05 Feb '10

Dear users, I'm doing some stress test with kamailio 1.5. When I'm at 50 CPS a see some strange messages: /usr/sbin/kamailio[12754]: CRITICAL:dialog:log_next_state_dlg: bogus event 2 in state 3 for dlg 0x2b0b911831b0 [3345:993496027] with clid '297-19089@IP-address' and tags '19089SIPpTag00297' '15519SIPpTag01304' Does anyone before experienced this before? Thanks Alex

1 0

[Kamailio-Users] Maximum retries exceed on transmission
by Jon Farmer 05 Feb '10

05 Feb '10

Hi I am getting this dreaded error on a asterisk box when call from one UA to another using Asterisk as a media relay and openser as a SBC and proxy. If I make a call from a UA to the PSTN through Asterisk all is fine. This only happens when calling from one UA to another on the same LAN. I have posted an example sip trace here http://pastebin.com/m2262b0ef My openser.cfg is here http://pastebin.com/d5e84aaf7 I realise this is a issue with Asterisk but I have not been able to find a fix. Therefore I figure there might be something I can do on openser. The problem seems to be with a ACK not being returned but my SIP knowledge is not good enough to know how to fix it. Any help is greatly appreciated. Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 5

[Kamailio-Users] new: interactive debugger for config
by Daniel-Constantin Mierla 05 Feb '10

05 Feb '10

Hello, the new debugger module can be used to do runtime debugging of SIP Router config file, in similar manner of gdb. The module is controlled via the RPC interface, therefore investigation can be done from a remote site. That makes it very handy to use with sercmd as well. One features is to print a log message for each action executed in the config for a SIP message, showing the execution path. The option can be enabled/disabled per process. The other one is step by step execution of config file. When a SIP message message comes it, SIP Router process will stop at first action and wait for commands, like: execute the action, evaluate a pseduo-variable, print to syslog a pseudo-variable, remove the breakpoints and continue execution without interruption, ... See the readme for more: http://sip-router.org/docbook/sip-router/branch/master/modules/debugger/deb… It is still some stuff to complete, some known issues to remove, but the module is usable and may help some of you to troubleshoot migrations from older versions to 3.0 (use master branch until your config is updated and running fine, then use it with stable 3.0). Hope is going to be useful and waiting for feedback to improve. Of course, contributions are most welcome! Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

4 5

[Kamailio-Users] Kamailio/SIPRouter Training
by Bruce McAlister 05 Feb '10

05 Feb '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, Are there any planned dates for Kamailio/SIPRouter training events around Europe in the next couple months. Looking forward to your replies..... Thanks Bruce -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJLa+kVAAoJEGJOkoTLLvBntMUIAJZEb7AaFDAdwKmTxlyokq30 A5eebX8mTUXk9fkzcHX0IoRRrl8eQKGqUbIl2312oQZtxJrffTksY3xiFFOdpBCp 78RDmKjLvmhAwq7jYPExk91BiV4no+dcAKuHsntwz8EfW9uHd34NRlp+TEc8Q2A8 QB7Ztfq8J+8KjGWJoRE27gqiHEIwZp7F7/rvn4N/VAuaTcaTtYhaiI52iIOA+aNM UTgkGMVzjjpPrwXh7C/DZERKThTifF01PzCx1DSlPiMczrGCGxHy8NEgAaHHsXb9 yrrtd/B4qWAbaYT8mRXisw9hNaGktdmCAA6Ksrq/xz/qjjVa84Tflo9d754PTGw= =kgI0 -----END PGP SIGNATURE-----

1 0

[Kamailio-Users] load_credentials problem
by Dubravko Caric 05 Feb '10

05 Feb '10

Hi all, i have a problem when using AVPs which are loaded on client registration by the "load_credentials". in the "subscriber" table i have "rpid" column (value 12345), "email address" (bob(a)example.com) and "acl" (value 3). when client registers, these three AVPs are loaded as you can see in debug. But when i want to use these AVPs it seems that they are empty. i don't understand if they are ready for use or do i need to load them and how? i'm not doing any actions on AVPs so i could by mistake delete or replace the original ones... version of kamailio is 1.5.1 Thanks Dubravko modparam("auth_db", "load_credentials", "$avp(i:66)=rpid;email_address;$avp(i:67)=acl") DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[0] at 0x783400 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783400)[0]=[password] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[1] at 0x783420 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783420)[1]=[acl] DBG:db_mysql:db_mysql_get_columns: use DB_INT result type DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[2] at 0x783440 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783440)[2]=[email_address] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[3] at 0x783460 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783460)[3]=[rpid] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type DBG:core:db_allocate_rows: allocate 16 bytes for rows at 0x783480 DBG:core:db_allocate_row: allocate 128 bytes for row values at 0x7834a0 DBG:core:db_str2val: converting STRING [1234] DBG:core:db_str2val: converting INT [3] DBG:core:db_str2val: converting STRING [bob(a)example.com] DBG:core:db_str2val: converting STRING [12345] DBG:auth_db:get_ha1: HA1 string calculated: 649766733fe475191f6c32d338aae51d DBG:auth:check_response: our result = 'f929dd9729d3c9447c4a4d632a762c85' DBG:auth:check_response: authorization is OK DBG:auth:post_auth: nonce index= 0 DBG:auth_db:generate_avps: set int AVP ""/67 = 3 DBG:auth_db:generate_avps: set string AVP "email_address"/0 = "bob(a)example.com" DBG:auth_db:generate_avps: set string AVP ""/66 = "12345" DBG:core:db_free_columns: freeing 4 columns ################# if (is_method("INVITE")) { xlog("L_INFO", "RPID AVP VALUE $avp(i:66)\n"); xlog("L_INFO", "EMAIL AVP VALUE $avp(s:email_address)\n"); xlog("L_INFO", "ACL AVP VALUE $avp(i:67)\n"); } ./kamailio[21687]: PRID AVP VALUE <null> ./kamailio[21687]: EMAIL AVP VALUE <null> ./kamailio[21687]: ACL AVP VALUE <null>

4 5

[Kamailio-Users] Kamailio v1.5.4 and v1.4.5 released
by Daniel-Constantin Mierla 05 Feb '10

05 Feb '10

Hello, Kamailio version 1.5.4 is out, built from latest svn version of SVN branch 1.5. All details about it are available at: http://www.kamailio.org/mos/view/News/NewsItem/Kamailio-v1.5.4-Released/ Also Kamailio v1.4.5 was released to mark the end of official maintenance for SVN branch 1.4. Details at: http://www.kamailio.org/mos/view/News/NewsItem/Kamailio-v1.4.5-Released/ If you are using a version from these release series, consider updating to the last one. Personally I recommend that you upgrade to the latest stable branch release v3.0.0 -- you get access lot of new features and improvements. Cheers, Daniel

3 5

Re: [Kamailio-Users] [kamailio-users] can't connect with remote database
by Chandrakant Solanki 05 Feb '10

05 Feb '10

On Thu, Feb 4, 2010 at 5:38 PM, Chandrakant Solanki < solanki.chandrakant(a)gmail.com> wrote: > > > On Thu, Feb 4, 2010 at 5:31 PM, marius zbihlei <marius.zbihlei(a)1and1.ro>wrote: > >> Chandrakant Solanki wrote: >> >>> Hi >>> >>> I have 3 server with 172.16.0.10/13/14 <http://172.16.0.10/13/14> IP >>> address. >>> >>> >>> My DB is located for 172.16.0.10. I tried to connect mysql from terminal, >>> it works great. >>> >>> But when I start kamailio it gives me following error.. >>> >>> Hello >> >> Just to double check Have you tried: >> >> mysql --user=openser -p --host=172.16.0.10 openser >> >> from 172.16.0.13 and it works? >> >> From localhost afaik mysql client might use Unix socks(/tmp/mysql.sock) . >> Use should double check that your MySQL server allows TCP connections >> >> Cheers >> >> Marius >> > > Hi mysql --user=openser -p --host=172.16.0.10 openser --socket=/var/lib/mysql/mysql. i have used this command from 172.16.0.13/14 it will works fine and connect to DB on 172.16.0.10 -- Regards, Chandrakant Solanki

1 0

[Kamailio-Users] OK, I've had enough "Ausente"
by Alex Balashov 04 Feb '10

04 Feb '10

Basta. Ya me has dicho que no estás. -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

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[Kamailio-Users] Openser with MySql Cluster
by Ricardo Coelho 04 Feb '10

04 Feb '10

Hi Right now I have a single openser server configured to work with a single mysql database. I want to enhance availability by adding another openser server and use Mysql Cluster. When failover occurs, despite the fact that he databases have the same data, on one of the softphones i receive a message saying "User not found" and on the other softphone i receive a message saying "Security check failed". Do I need to change something in openser's configuration to make it work with mysql cluster? Thanks in advanced

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Re: [Kamailio-Users] break a call?
by BERGANZ Francois 04 Feb '10

04 Feb '10

Where can I activate it? -----Message d'origine----- De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Envoyé : jeudi 4 février 2010 14:09 À : BERGANZ Francois Objet : Re: [Kamailio-Users] break a call? probably the problem is that you can not use tm module functions in onreply_route. You have to handle the responses in the failure route (every final response >=300 triggers the failure_route (if activated)). regards klaus Am 04.02.2010 13:40, schrieb BERGANZ Francois: > Line 254 is t_reply("406", "Not Acceptable"); > > > > -----Message d'origine----- > De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Envoyé : jeudi 4 février 2010 13:08 > À : BERGANZ Francois > Objet : Re: [Kamailio-Users] break a call? > > > > Am 04.02.2010 12:37, schrieb BERGANZ Francois: >> I have that error: >> >> >> onreply_route[1] { >> if(t_check_status("3[0-9][0-9]")){ >> xlog("L_ERR", "3XX!!!!!!!!!!!\n"); >> t_reply("406", "Not Acceptable"); >> exit; >> } >> } >> >> Feb 4 12:36:41 [4584] CRITICAL:core:yyerror: parse error in config >> file, line 254, column 16-17: Command cannot be used in the block > > How will I know what line 254 is? > > The code I sent uses the failure-route, not the reply-route. > > klaus > > >> >> >> Any idea? >> >> Thank you >> >> >> >> -----Message d'origine----- >> De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >> Envoyé : jeudi 4 février 2010 08:59 >> À : BERGANZ Francois >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] break a call? >> >> e.g.: >> >> failure_route[1] { >> if(t_check_status("302")) { >> xlog("L_INFO","302 redirect received\n"); >> t_reply("403","received 302 from downstream client"); >> exit; >> } >> } >> >> klaus >> >> Am 04.02.2010 08:37, schrieb BERGANZ Francois: >>> Hello, >>> >>> I know that we can break a call with sl_reply(). >>> >>> But, in the on_reply route, how can I break a call if I receive a >>> 3XX (I cant use a sl_reply() function!) ? >>> >>> Thank you >>> >>> >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >

2 1

Routing Problems (Sorry, Text file is now included)
by Andreas Rehbein 04 Feb '10

04 Feb '10

Hello, we use Kamailio 3.0 on a Red Hat EL5.4 system with openssl 0.9.8e (the current Red Hat OpenSSL version). We want to use Kamailio 3.0 in front of our Call Server (OpenSER 1.3.2) as a security gateway. So the Call Server should only deal tcp, while the Security Gateway terminates the TLS Data which he receives from the User Agents but forwards the SIP-Messages via TCP to the Call Server. UA ---SIP_over_TLS---> Security Gateway (Kam3.0) ---SIP_over_TCP---> Call Server (OpenSER1.3.2) UA<---SIP_over_TLS--- Security Gateway (Kam3.0) <---SIP_over_TCP--- Call Server (OpenSER1.3.2) Nearly everything works fine in this scenario: User Agents are able to register and when they are sending INVITES the callee receives it. The problem we have right now is: if Kamailio 3.0 receives BYE we get "477 Unfortunatly error on sending to next hop occured". It seems that the messages are ok... I attached a text file with the BYE message and the errors. Explanation for the text file: * UA1: 192.168.0.126 * UA2: 192.168.0.176 *Security Gateway (Kam3.0): 192.168.0.89 *Call Server (OpenSER1.3.2): 192.168.0.106 No. Time Source Destination Protocol Info 101 7.964015 192.168.0.106 192.168.0.89 SIP Request: BYE sip:+497118113608227@192.168.0.89:37260;transport=TLS Frame 101 (1138 bytes on wire, 1138 bytes captured) Ethernet II, Src: AlliedTe_3e:b8:05 (00:15:77:3e:b8:05), Dst: Dell_a0:e2:3f (00:1c:23:a0:e2:3f) Internet Protocol, Src: 192.168.0.106 (192.168.0.106), Dst: 192.168.0.89 (192.168.0.89) Transmission Control Protocol, Src Port: 50445 (50445), Dst Port: sip (5060), Seq: 1234, Ack: 1, Len: 1072 Session Initiation Protocol Request-Line: BYE sip:+497118113608227@192.168.0.89:37260;transport=TLS SIP/2.0 Message Header Record-Route: <sip:192.168.0.106;transport=tcp;lr=on;ftag=ba3448a0> Record-Route: <sip:192.168.0.89;transport=tcp;r2=on;lr=on> Record-Route: <sip:192.168.0.89:5061;transport=tls;r2=on;lr=on> Via: SIP/2.0/TCP 192.168.0.106;branch=z9hG4bK854f.b305c913.0;i=19 Via: SIP/2.0/TCP 192.168.0.89;rport=37260;branch=z9hG4bK854f.dac34f77.0;i=a Via: SIP/2.0/TLS 192.168.0.126:62399;branch=z9hG4bK-d8754z-51f50d6cfa997da7-1---d8754z-;rport =1876 Max-Forwards: 68 Route: <sip:192.168.0.89;transport=tcp;r2=on;lr=on> Route: <sip:192.168.0.89:5061;transport=tls;r2=on;lr=on> Contact: <sip:+497118113608226@192.168.0.89:37260;transport=TLS;rinstance=972e46956fc 87f26> To: "+497118113608227"<sip:+497118113608227@192.168.0.106>;tag=15c25277 From: <sip:+497118113608226@192.168.0.106>;tag=ba3448a0 Call-ID: ZjdiMWY5NTYxMTMyN2Q5NTkwOWVmZWExOTEwMDhjYzQ. CSeq: 2 BYE User-Agent: Bosch Communicator release 2.5.4.2 GAC4 stamp 54836 Reason: SIP;description="User Hung Up" Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 103 7.964791 192.168.0.89 192.168.0.106 SIP Status: 477 Unfortunately error on sending to next hop occurred (477/SL) Frame 103 (887 bytes on wire, 887 bytes captured) Ethernet II, Src: Dell_a0:e2:3f (00:1c:23:a0:e2:3f), Dst: AlliedTe_3e:b8:05 (00:15:77:3e:b8:05) Internet Protocol, Src: 192.168.0.89 (192.168.0.89), Dst: 192.168.0.106 (192.168.0.106) Transmission Control Protocol, Src Port: sip (5060), Dst Port: 50445 (50445), Seq: 1, Ack: 2306, Len: 821 Session Initiation Protocol Status-Line: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/SL) Message Header Via: SIP/2.0/TCP 192.168.0.106;branch=z9hG4bK854f.b305c913.0;i=19 Via: SIP/2.0/TCP 192.168.0.89;rport=37260;branch=z9hG4bK854f.dac34f77.0;i=a Via: SIP/2.0/TLS 192.168.0.126:62399;branch=z9hG4bK-d8754z-51f50d6cfa997da7-1---d8754z-;rport =1876 To: "+497118113608227"<sip:+497118113608227@192.168.0.106>;tag=15c25277 From: <sip:+497118113608226@192.168.0.106>;tag=ba3448a0 Call-ID: ZjdiMWY5NTYxMTMyN2Q5NTkwOWVmZWExOTEwMDhjYzQ. CSeq: 2 BYE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 [truncated] Warning: 392 192.168.0.89:5060 "Noisy feedback tells: pid=10378 req_src_ip=192.168.0.106 req_src_port=50445 in_uri=sip:+497118113608227@192.168.0.89:37260;transport=TLS out_uri=sip:+497118113608227@192.168.0.89:37260;transport No. Time Source Destination Protocol Info 104 7.964845 192.168.0.89 192.168.0.106 SIP Status: 477 Unfortunately error on sending to next hop occurred (477/TM) Frame 104 (887 bytes on wire, 887 bytes captured) Ethernet II, Src: Dell_a0:e2:3f (00:1c:23:a0:e2:3f), Dst: AlliedTe_3e:b8:05 (00:15:77:3e:b8:05) Internet Protocol, Src: 192.168.0.89 (192.168.0.89), Dst: 192.168.0.106 (192.168.0.106) Transmission Control Protocol, Src Port: sip (5060), Dst Port: 50445 (50445), Seq: 822, Ack: 2306, Len: 821 Session Initiation Protocol Status-Line: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Message Header Via: SIP/2.0/TCP 192.168.0.106;branch=z9hG4bK854f.b305c913.0;i=19 Via: SIP/2.0/TCP 192.168.0.89;rport=37260;branch=z9hG4bK854f.dac34f77.0;i=a Via: SIP/2.0/TLS 192.168.0.126:62399;branch=z9hG4bK-d8754z-51f50d6cfa997da7-1---d8754z-;rport =1876 To: "+497118113608227"<sip:+497118113608227@192.168.0.106>;tag=15c25277 From: <sip:+497118113608226@192.168.0.106>;tag=ba3448a0 Call-ID: ZjdiMWY5NTYxMTMyN2Q5NTkwOWVmZWExOTEwMDhjYzQ. CSeq: 2 BYE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 [truncated] Warning: 392 192.168.0.89:5060 "Noisy feedback tells: pid=10378 req_src_ip=192.168.0.106 req_src_port=50445 in_uri=sip:+497118113608227@192.168.0.89:37260;transport=TLS out_uri=sip:+497118113608227@192.168.0.89:37260;transport Do you have any suggestions? Thank you very much in advance! Regards Andreas

1 0

[Kamailio-Users] Sip MESSAGE
by Abdul Hakeem 04 Feb '10

04 Feb '10

Hello, Does anyone knows if Kamalito supports MESSAGE method & where I can find the documentation or API's ? Regards, Abdul Hakeem

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[Kamailio-Users] new module: mqueue
by Elena-Ramona Modroiu 04 Feb '10

04 Feb '10

Hi, few details about the new module mqueue. It provides a generic message queue for SIP Router inter process communication. Items in the queue are pairs of (key, value), both string. The items are consumed in the order they were added in the queue (first in first out). More details at: http://sip-router.org/docbook/sip-router/branch/master/modules/mqueue/mqueu… http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#mqueue_module_p… The module is handy to use for time consuming and/or blocking operations (eg, email notifications). Not to hold the SIP worker process, you can add an item in a queue that is going to be consumed by another process (like timer processes created via rtimer module). Best regards, Ramona

1 0

[Kamailio-Users] [kamailio-users] can't connect with remote database
by Chandrakant Solanki 04 Feb '10

04 Feb '10

Hi I have 3 server with 172.16.0.10/13/14 IP address. My DB is located for 172.16.0.10. I tried to connect mysql from terminal, it works great. But when I start kamailio it gives me following error.. *kamailio.cfg* loadmodule "usrloc.so" ----- ----- ----- modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@172.16.0.10/openser ") *error.... *Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:core:db_do_init: could not add connection to the pool Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:usrloc:child_init: child(8): failed to connect to database Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:core:init_mod_child: failed to initializing module usrloc, rank 8 Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:core:main_loop: init_child failed for UDP listener Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28793]: ERROR:db_mysql:db_mysql_new_connection: driver error: Can't connect to MySQL server on '172.16.0.10' (4)* * -- Regards, Chandrakant Solanki

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[Kamailio-Users] break a call?
by BERGANZ Francois 04 Feb '10

04 Feb '10

Hello, I know that we can break a call with sl_reply(). But, in the on_reply route, how can I break a call if I receive a 3XX (I cant use a sl_reply() function!) ? Thank you

2 2

[Kamailio-Users] xlog enhancements
by Elena-Ramona Modroiu 04 Feb '10

04 Feb '10

Hi, I added two new functions to xlog (modules_k): - xlogl(level, message) - xdbgl(message) They are similar to the existing functions, in addition they print the config line at the beginning of the message. The printed value is: "cfglineno:message" In this way you can spot easily in the logs where the function was used to print a particular message. http://sip-router.org/docbook/sip-router/branch/master/modules_k/xlog/xlog.… Regards, Ramona

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[Kamailio-Users] Loose Route / Re-Invite
by Brandon Armstead 04 Feb '10

04 Feb '10

Hello everyone, I'm just curious as to see what some of you guys do in regards to handling a Re-Invite that comes back downstream to a NATTED UAC. For example, call scenario: UAC -> Kamailio (Fix Nated Contact) -> PSTN Re-Invite Occurs: PSTN -> Kamailio -> UAC UAC (200 OK w/ NAT RFC1918 contact) -> Kamailio (branch flags at this point are not notifying of NAT, due to the downstream direction of the INVITE, so RFC1918 address exists, but does not fix_nated_contact) -> PSTN PSTN does not appropriately ACK. So, what are your guys solutions for solving this problem? Is the best way to add an attribute onto the contact header sent in original INVITE? Are there other ways of handling? What is the best, cleanest method. Possible to handle with AVP's? Thanks in advanced for all of your input! Sincerely, Brandon

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[Kamailio-Users] [OT] acabrera@sintys.gov.ar is away, did you know?
by Iñaki Baz Castillo 04 Feb '10

04 Feb '10

Hi, am I the only person receiving an auto-reply mail from acabrera(a)sintys.gov.ar with subject "Ausente" ("Away") for each mail sent to this list (and perhaps to others)? I've already sent him a direct mail. However, why do people use such auto- reply stuf in mail?? this is mail, an asynchronous communication, the receipt can read the mail whenever he prefers, not need to tell me "I'm away now". grrrr -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] Loose Route / Re-Invite (brandon@cryy.com)
by Brandon Armstead 03 Feb '10

03 Feb '10

Hello, Please refer to sip trace: http://pastebin.org/86010. On LINE 294, PSTN sends the Re-INVITE On LINE 378, REGISTRAR sends 200 OK to SIP SBC (with RFC 1918 Address, that is being relayed from the UAC that also sends this same address upon 200 OK of the Re-Invite). Let me know if this makes sense, this trace was taken on the SBC.

4 5

[Kamailio-Users] Registration delays when using Kamailio proxy
by Vikram Ragukumar 03 Feb '10

03 Feb '10

Hello, I am using an x-lite softphone to register with my SIP domain, (a) Without an outbound proxy - Registration is successful and takes about 3 seconds (b) With an outbound proxy - Registration is successful but takes nearly 15 seconds I briefly describe below the sequence of SIP messages while using the outbound proxy Register softphone -> proxy Register proxy -> SIP server Register softphone -> proxy 407 Proxy authentication reqd. SIP server -> proxy Register softphone -> proxy 407 Proxy authentication reqd. proxy -> softphone Register softphone -> proxy (With Digest response) Register proxy -> SIP server (With Digest response) 200 OK Sip server -> proxy Register softphone -> proxy (With Digest response) I am attaching a log with wireshark captures that show SIP messages exchanged between the SIP server,SIP proxy and the x-lite softphone. The log shows several messages being duplicated (REGISTER, 200 OK) 1) Why does the softphone continue to send REGISTER messages after receiving a 200 OK response ? (almost 15 repetitions of REGISTER after receiving 200 OK) 2) What is the reason for the delay in registration times with/without the proxy ? Thanks and Regards, Vikram. No. Time Source Destination Protocol Info 1 0.000000 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 1 (611 bytes on wire, 611 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 3 0.839564 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 3 (722 bytes on wire, 722 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 4 0.952097 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 4 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100052 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605203001203402405205305" Content-Length: 0 No. Time Source Destination Protocol Info 5 1.404504 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 5 (722 bytes on wire, 722 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 6 1.517292 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 6 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100053 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605303001203402405305305" Content-Length: 0 No. Time Source Destination Protocol Info 7 1.522829 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 7 (611 bytes on wire, 611 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 8 2.522300 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 8 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100052 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605203001203402405205305" Content-Length: 0 No. Time Source Destination Protocol Info 9 2.730896 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 9 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 10 2.796109 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 10 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100052 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605203001203402405205305" Content-Length: 0 No. Time Source Destination Protocol Info 11 3.308767 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 11 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 12 4.099907 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 12 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 13 4.216914 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 13 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100055 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605503001203402405505305" Content-Length: 0 No. Time Source Destination Protocol Info 14 4.324392 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 14 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 15 4.704708 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 15 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 16 4.822244 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 16 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100056 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605603001203402405605305" Content-Length: 0 No. Time Source Destination Protocol Info 17 5.710292 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 17 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100055 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605503001203402405505305" Content-Length: 0 No. Time Source Destination Protocol Info 18 5.724689 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 18 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100055 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605503001203402405505305" Content-Length: 0 No. Time Source Destination Protocol Info 19 5.832474 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 19 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 21 7.455368 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 21 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 22 7.457267 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 22 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 23 7.574533 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 200 OK (1 bindings) Frame 23 (539 bytes on wire, 539 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 3 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100059 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=60 Expires: 60 Content-Length: 0 No. Time Source Destination Protocol Info 24 8.355247 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 24 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 25 8.472891 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 200 OK (1 bindings) Frame 25 (539 bytes on wire, 539 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 3 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030241100000 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=60 Expires: 60 Content-Length: 0 No. Time Source Destination Protocol Info 26 9.310268 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 200 OK (1 bindings) Frame 26 (476 bytes on wire, 476 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 3 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100059 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=60 Expires: 60 Content-Length: 0 No. Time Source Destination Protocol Info 29 10.939788 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 29 (796 bytes on wire, 796 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-075267174a75f03a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=0 To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 30 11.097051 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 30 (907 bytes on wire, 907 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bKe0c8.73270ff2.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-075267174a75f03a-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=0 To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound

4 5

[Kamailio-Users] protocol/port mismatch on BYE from Calle to Caller
by Arafath-uz-zaman khan 03 Feb '10

03 Feb '10

Hello All, I need a working example for sending all request to another sip server using kamailio. X-lite-->TLS(7161)-->Kamailio-->UDP(5060)-->SIP Server-->PSTN Number. i have attached my cfg file. The problem is when i dialed any pstn number everything working fine except the by msg. If the caller send BYE callee get hangup but if Calle send BYE caller never hangup the call. i got the following msg on my log file Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_msg: method: <BYE> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_msg: uri: <sip:91131@192.168.10.2:5060> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_msg: version: <SIP/2.0> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=2 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: cseq <CSeq>: <1> <BYE> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bk010213101521011481381407>; state=16 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_via: end of header reached, state=5 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: via found, flags=2 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: this is the first via Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:receive_msg: After parse_msg... Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:receive_msg: preparing to run routing scripts... Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=100 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to_param: tag=b23c1531 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: end of header reached, state=29 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: display={"91131"}, ruri={sip:91131@192.168.10.2:5060} Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: <To> [51]; uri=[sip:91131@192.168.10.2:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: to body ["91131"<sip:91131@192.168.10.2:5060>] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: content_length=0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: found end of header Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:maxfwd:is_maxfwd_present: max_forwards header not found! Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:check_via_address: params 192.168.10.2, 192.168.10.2, 0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=80 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:siputils:has_totag: totag found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=200 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:is_preloaded: is_preloaded: No Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 12==13 && [192.168.10.2] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 12==13 && [192.168.10.2] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 7161 matches port 5060 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: no match for: [192.168.10.2:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_aliases: no match for: [0:192.168.10.2:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:check_self: host != me Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 13==13 && [192.168.10.1] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: match found for: [192.168.10.1:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:check_self: host == me Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:after_loose: Topmost route URI: 'sip:192.168.10.1;r2=on;lr=on;nat=yes' is me Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 13==13 && [192.168.10.1] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 7161 matches port 7161 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: match found for: [192.168.10.1:7161] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=200 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: found end of header Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:find_next_route: No next Route HF found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:after_loose: no next URI found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:check_route_param: params are <;r2=on;lr=on;nat=yes> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=40 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to_param: tag=01021310150514813644546133 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: end of header reached, state=29 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: display={"911317434"}, ruri={sip:911317434@192.168.10.1:5060} Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:siputils:has_totag: totag found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=78 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_lookup_request: start searching: hash=850, isACK=0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_lookup_request: proceeding to pre-RFC3261 transaction matching Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_lookup_request: no transaction found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:run_reqin_callbacks: trans=0xb615b5f8, callback type 1, id 0 entered Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=78 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:mk_proxy: doing DNS lookup... *Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_send_socket: force_send_socket of different proto (1)! Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: WARNING:core:get_send_socket: protocol/port mismatch* Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:set_timer: relative timeout is 500000 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:insert_timer_unsafe: [4]: 0xb615b744 (46600000) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:set_timer: relative timeout is 30 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:insert_timer_unsafe: [0]: 0xb615b760 (76) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_relay_to: new transaction fwd'ed Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:destroy_avp_list: destroying list (nil) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:receive_msg: cleaning up Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:utimer_routine: timer routine:4,tl=0xb615b744 next=(nil), timeout=46600000 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb615b5f8, BYE sip:9 ... ) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:set_timer: relative timeout is 1000000 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:insert_timer_unsafe: [5]: 0xb615b744 (47600000) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:utimer_routine: timer routine:5,tl=0xb615b744 next=(nil), timeout=47600000 Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb615b5f8, BYE sip:9 ... ) Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:set_timer: relative timeout is 2000000 Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:insert_timer_unsafe: [6]: 0xb615b744 (49600000) Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : done

2 3

[Kamailio-Users] the best editor?
by BERGANZ Francois 03 Feb '10

03 Feb '10

Hello all, what is the best edit tool to see quickly the right syntax ... ? which language to choose?

5 6

[Kamailio-Users] KAMAILIO 1.4.3 no TLS Stop processing calls when Radius server goes down
by Omar 03 Feb '10

03 Feb '10

I have set it up the Radius for Kamailio, it works fine, until the Radius server goes down it did not go down for a while, when tis happes the Kamailio server stops processing any calls. I do not think is a normal behaviour, probably a bug, can anybody please tell me if this is expected behave in kamailio. or was already noticed before and was already fixed in a later release. Is possible to point to more than 2 radius servers in parallel? NOTE: i was running radius and DB in parallel with no issues, until the RADIUS server goes down. __________________________ O

3 2

[Kamailio-Users] dialog and pua_dialoginfo: Issue regarding vanished PUBLISH messages
by Martin Koenig 03 Feb '10

03 Feb '10

Hello all, I've run into an issue of Mar 2009: http://lists.kamailio.org/pipermail/users/2009-March/022143.html This seems to be unresolved in v1.5.4, the then presented work-around via modparam("pua_dialoginfo", "override_lifetime", 600) still works fine. The question is whether this should work differently nowadays, or if notice should be put into the docs? Best regards, Martin

2 2

[Kamailio-Users] alias_db: append_branches and dialog
by Martin Koenig 03 Feb '10

03 Feb '10

Hello all, two questions regarding alias_db: 1) How do I make use of the append_branches function of alias_db? Trying to insert two or more aliases is not possible in the current db layout ( UNIQUE KEY `alias_idx` (`alias_username`,`alias_domain`) ). The idea was to enable simple db-based parallel branching ("group call") via alias_db by inserting one or more users to one alias. This seems not possible. 2) Use of alias_db "breaks" presence_dialog, because dialog matching is based on from / to header (instead from / final AoR after alias_db_lookup() ), thus presence_dialog events cannot be created and PUBLISHed to end devices. I was wondering whether it would be sufficient to uac_replace_from in branch_route[] before the final t_relay work around this, or if there is another solution out there? Many thanks and best regards, Martin

2 1

Re: [Kamailio-Users] append_rpid_hf add privacy?
by Daniel-Constantin Mierla 03 Feb '10

03 Feb '10

Hello, On 2/2/10 1:18 PM, alex pappas wrote: > Hi Daniel, > > I'm capturing with tcpdump the packets when they live the server and I > don't see the extra header added. > are you having the rpid value in subscriber table? Is it loaded via load_credentials of auth_db or do you set the right avp in script? Have you tried append_hf()? Seems you want to add r-uri to rpid, which is easier via append_hf() then. Cheers, Daniel > Cheers > Alex > > > > On Tue, Feb 2, 2010 at 12:24 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hi Alex, > > > On 2/2/10 11:04 AM, alex pappas wrote: > > Dear friends, > > I'm trying to give to a user the option to have privacy in a > call. I'm trying to do that by adding an extra header with: > append_rpid_hf("sip", > ";party=calling;id-type=subscriber;privacy=off;screen=no"); > > I'm using this header in the begining of the kamailio.cfg > > if(($rU=="99999") || ($rU=="77777")) > { > append_rpid_hf("sip", > "$ru;party=calling;id-type=subscriber;privacy=off;screen=no"); > xlog("alx ------- We have rU match > --------"); > } > > When I see the sip messages I don't see this extra header > added into it. > > Does anyone had experience in this ? > > the header is visible when the message is forwarded. To see it > immediately, but only in 3.0.0, you can use msg_apply_changes(). > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > eLearning class for Kamailio 3.0.0 > Starting Feb 8, 2010 > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

2 4

[Kamailio-Users] preparing kamailio 1.5.4 and 1.4.5 releases
by Daniel-Constantin Mierla 02 Feb '10

02 Feb '10

Hello, soon I will start preparing the 1.5.4 and 1.4.5. The only issue reported and not clarified yet is the tcp failover that I tried to reproduce today without luck, might be dependent of (triggered faster on) particular architecture. It can be fixed once gathering more details or being able to reproduce. If you are aware of something else, please report to devel list. 1.4.5 will mark the official support for 1.4.x series, although some fixes may go back there from time to time, but I don't think will be another release of that branch. I recommend to update to newer version or stick to svn for updates. Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

1 0

[Kamailio-Users] PV bug in modules_k/acc/acc_extra.c in K 3.0.0
by Alex Balashov 02 Feb '10

02 Feb '10

I ran into a very interesting problem last night with 'acc' that took me a few hours to solve, so I am posting my workaround for collective benefit. It is probably not the cleanest and most appropriate solution because I am not extensively familiar with the 'acc' code; this is my first dive into it. Thus, I do not presume to submit an actual patch. The problem I had was that PVs read from the db_extra parameter that had an internal type of integer (TYPE_INT) because of their lexical scope were not being correctly evaluated, resulting in bizarre values being put in the database. I am using K 3.0.0 on 64-bit Xen guest running CentOS with db_postgres. I did not check to see if this problem occurs with db_mysql as I do not have MySQL anywhere nearby, but I don't believe it is happening at the database layer anyhow. I also know this problem was not present in 1.5.x, or several of my installations would be non-functional; this must have something to do with use of new 'srdb1' stuff in the database side of 'acc', I hypothesise. Anyway, the actual problem went like this. Let us say, for simplicity of example, that I am using 'acc' with Postgres backing and have something like this: modparam("acc", "db_extra", "account_id=$avp(s:account_id)") ... modparam("acc", "db_flag", 1) What was happening was that if I did this in a request route: $avp(s:account_id) = 42; ... setflag(1); if(!t_relay()) sl_reply_error(); exit; Very strange and occasionally bizarre values, mostly duplicates of other values I have in db_extra, would be written to acc.account_id (a column of type 'integer') - at any rate, certainly not 42. I enabled SQL statement logging on my PostgreSQL database to ensure that the data being inserted was literally not 42, rather than this being some sort of database-layer type conversion / representation problem. Sure enough, it was true, although I also did notice that 'acc' encloses all inserted values in single quotes to be treated as string literals. But this is immaterial, because RDBMs automatically cast such input: INSERT INTO acc (..., account_id) VALUES(..., '12'); Now, I could fix the problem by simply assigning the AVP a string literal instead: $avp(s:account_id) = '42'; Now the data was inserted normally. I also tested with user variables ($var(...)) and the same was true, so the problem is not AVP-specific. Also, all other evaluative contexts for PVs with these values work fine, such as printing them. The problem does not seem to be with 'db_postgres', neither with 'pv', nor with 'avpops' -- it is particular to 'acc'. After some hours of investigation I found that what was happening is that the type of the PV being referenced is not being correctly packed into the various arrays populated by extra2strar() in modules_k/acc/acc_extra.c, and thus was being misread when iterated in acc_db_request() in modules_k/acc/acc.c. Specifically, this check was inadequate: if (value.rs.s+value.rs.len==static_detector) { val_arr[n].s = int_buf + r*INT2STR_MAX_LEN; val_arr[n].len = value.rs.len; memcpy(val_arr[n].s, value.rs.s, value.rs.len); r++; LM_INFO("New int tenant: %s\n", val_arr[n].s); } else { val_arr[n] = value.rs; LM_INFO("New str tenant: %s\n", value.rs); } In my debugging, all PV values were showing up as "New str tenant" regardless of whether they were natively integers, and regardless of what the PV spec type was actually set to. The type was set correctly on the PV spec, so of course the integer value got stored correctly in int_var[] below: if (value.flags&PV_VAL_INT) { int_arr[n] = value.ri; type_arr[n] = TYPE_INT; } else { type_arr[n] = TYPE_STR; } ... but since 'acc' wants to treat all extra values as strings, this was not sufficient, because what was failing was this condition: if (value.rs.s+value.rs.len==static_detector) I guess value.rs.s in case of the PV value being PV_VAL_INT is not null terminated properly? This I truly do not know. In any event, I just added a check on the PV flags to see if it is an integer value and this fixed my problem: if (value.rs.s+value.rs.len==static_detector || value.flags & PV_VAL_INT) { ... however, I do not know that this was the most insightful solution. This is rather critical I think, because it means anyone trying to store PV values that are initialised with a natively numerical type in script (or extracted as a numerical value from a database, e.g. sql_query("SELECT account_id::integer FROM table ..."), I found this to be a problem as well) cannot store them in the database reliably using 'db_extra'. Thanks! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 3

[Kamailio-Users] isup in invite
by Eric Hiller 02 Feb '10

02 Feb '10

I have an issue where my sip endpoint is sending ISUP information in the invite. IE: sip:+1xxxxxxxxxx;npdi=yes;cic=+1-0110@xxxx:5060 and when I run the enum query I am getting: 0(24410) enum_query(): uri user is not an E164 number This is most likely due to the fact that $rU=+17794294010;npdi=yes;cic=+1-0110 Does openser not read the user properly? Is there a way to correct this? I am running openser 1.2.2 notls Thanks for any help! -Eric _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/196390709/direct/01/

6 24

[Kamailio-Users] append_rpid_hf add privacy?
by alex pappas 02 Feb '10

02 Feb '10

Dear friends, I'm trying to give to a user the option to have privacy in a call. I'm trying to do that by adding an extra header with: append_rpid_hf("sip", ";party=calling;id-type=subscriber;privacy=off;screen=no"); I'm using this header in the begining of the kamailio.cfg if(($rU=="99999") || ($rU=="77777")) { append_rpid_hf("sip", "$ru;party=calling;id-type=subscriber;privacy=off;screen=no"); xlog("alx ------- We have rU match --------"); } When I see the sip messages I don't see this extra header added into it. Does anyone had experience in this ? Thank you Alex

2 3

[Kamailio-Users] new feature: remote user registration
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, if you follow the sr-dev mailing list, you may have noticed some new features added in master branch (for the 3.1.0). I will send more details about each, now: remote user registration. The uac module can register in behalf of users to remote servers. The user details (credentials, local and remote username/domain) are loaded from database. The module sends registration at start-up and then refreshes just before expiration time. If the REGISTER is challenged for authentication, uac will re-send the REGISTER with proper credentials. A typical scenario is when users have accounts to many voip providers and they want to receive calls to those accounts via your sip server (inbound traffic). Of course, the user has to give details such as username/password to you. Other use case can be dynamic interconnection/peering. you can imagine a scalable solution where you have a traffic dispatcher and worker servers. When an worker becomes active, it registers to traffic dispatcher for receiving traffic... and examples can continue. Hope will help you in various cases! Readme at: http://sip-router.org/docbook/sip-router/branch/master/modules_k/uac/uac.ht… SQL to create the mysql table is in utils/kamctl/mysql/uac-create.sql Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

4 4

[Kamailio-Users] new module: mtree
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, the master branch got a new module named mtree. It is able to load database records in the form of (key,value) and index them in shared memory trees based on key. One db table is loaded in one mtree, you can define as many trees as you need. The typical use case is to load arbitrary data related to DIDs or prefixes, offering very fast lookup even with large number of records. The value can be string or digested in a runtime structure, so far being able to digest dstid:weight values, useful for prefix matching on weight rather than longest match. The trees are read only, with option to reload via mi command (rpc to be added soon). Online readme at: http://sip-router.org/docbook/sip-router/branch/master/modules/mtree/mtree.… Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

1 0

[Kamailio-Users] kamailio 1.4 cpc
by Asim Riaz 01 Feb '10

01 Feb '10

Hi List, Is it possible to add diffrent CPC(calling party catagory) for diffrent type of the number in the INVITE before sending out to an upstream SIP server? INVITE ----> Kamailio(add CPC) -------> SIP_Proxy_2 Thanks, Asim

3 9

[Kamailio-Users] Fosdem 2010 - reminder
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, a reminder for those that already received (also sorry for cross-posting) and invitation for new ones, Fosdem 2010 happens this weekend in Brussels and we will have the traditional SIP routing dinner on Saturday Feb 6, trying to get all of us up-to-date regarding projects such as ser, kamailio (openser), sip-router, sems, asterisk, freeswitch, a.s.o. If you plan to attend, please confirm till Wednesday (send me an email), we need to have a fair estimation of attendees number to make the restaurant reservation. See more at: http://sip-router.org/2010/01/31/fosdem-2010-meeting-in-brussels/ See you at Fosdem, Daniel -- Daniel-Constantin Mierla eLearning Class for Kamailio 3.0.0 Starts Feb 8, 2010 * http://www.asipto.com/

1 0

[Kamailio-Users] project updates
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, I do not know how many noticed so far, for several days the management web page was update to reflect an enlarged group of people that represent Kamailio project. Since the collaboration within SIP Router project proved fruitful, it is clear that the future development is related to that project, therefore devel decisions will happen there (join sr-dev(a)lists.sip-router.org to be in touch). The last year and half was filled with heavy integration work and testing, most of it not very challenging, but had to be done, so we wanted to acknowledge the help given by our most active community contributors. As a project policy, we maintain officially the latest two stable releases, now being 1.5.x and 3.0.x. The efforts of the group will be directed in advocating and promoting Kamailio and SIP Router projects world wide. Also some of them sponsor and take care of maintaining resources such as servers, dns and web pages. http://www.kamailio.org/mos/view/Management/ Given the big amount of work done for 3.0.0, the feedback so far is very good, in these tree weeks we discovered few issues, most relevant affecting sip_trace and uac modules, with a pretty active discussions pool about this version. 3.0.0 branches are run by some open voip providers for quite some time now (e.g., voipuser.org and iptel.org), servicing thousands of active users, ensuring that the core modules (e.g., acc, tm, auth, nat traversal, group, dialog, presence, dispatcher, pv, textops, ...) are working smooth. As a side note, I have it in other several production systems. While I encourage everyone to send feedback about 3.0.0, worth to mention that devel version (git master branch to become 3.1.0) has already quite a lot of new features, so if you are looking for adventures or need Lua API, geoip location, in memory tree indexed data, Python API, user registrations to remote SIP servers (for inbound calls), check-out git master and give it a try. Note that master is as stable as 3.0.0 branches so far regarding old code, new additions haven't touched other parts, being stand alone. Looking forward to 3.1.0 as introducing new features but also providing more tight integration with SIP Router, especially on some common used modules. Cheers, Daniel

1 0

[Kamailio-Users] help me to find the problem
by shaon＠opennets.net 31 Jan '10

31 Jan '10

Hello I am new with kamailio. here is my problem X-lite -->TLS-->Kamailio(TLS+RegisterForwrd)-->UDP-->RegisterServer-->PSTN/VOIPCall I am able to forward my register request to the register server, and make call. Able to talk from both side. But when i hangup the call from termination side call never hangup from origination side. I attached my config file. I got following log on my kamailio.log file ############# Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_lookup_request: start searching: hash=53225, isACK=0 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_lookup_request: proceeding to pre-RFC3261 transaction matching Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_lookup_request: no transaction found Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:run_reqin_callbacks: trans=0xb61945d0, callback type 1, id 0 entered Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:mk_proxy: doing DNS lookup... Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:get_send_socket: force_send_socket of different proto (1)! Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: WARNING:core:get_send_socket: protocol/port mismatch Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:set_timer: relative timeout is 500000 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:insert_timer_unsafe: [4]: 0xb619471c (40400000) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:set_timer: relative timeout is 30 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:insert_timer_unsafe: [0]: 0xb6194738 (69) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_relay_to: new transaction fwd'ed Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:destroy_avp_list: destroying list (nil) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:receive_msg: cleaning up Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:utimer_routine: timer routine:4,tl=0xb619471c next=(nil), timeout=40400000 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb61945d0, BYE sip:9 ... ) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:set_timer: relative timeout is 1000000 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:insert_timer_unsafe: [5]: 0xb619471c (41400000) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:retransmission_handler: retransmission_handler : done Jan 31 15:54:46 ser /usr/local/sbin/kamailio[7581]: DBG:tm:utimer_routine: timer routine:5,tl=0xb619471c next=(nil), timeout=41400000 Jan 31 15:54:46 ser /usr/local/sbin/kamailio[7581]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb61945d0, BYE sip:9 ... ) Can anybody help me to solve the issue? Shaon

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[Kamailio-Users] ERROR:tm:w_t_reply failed to send a t_reply to a message for which no transaction-ate has been established
by AnthonyD 31 Jan '10

31 Jan '10

Dear List, I am running Kamailio (1.5.3-notls (i386/linux)) with the following modules: loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "lcr.so" loadmodule "dialog.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "domain.so" I get a lot of these errors in my logs: ERROR:tm:w_t_reply failed to send a t_reply to a message for which no transaction-ate has been established Can someone give me some info on these errors and how to trouble shoot the cause? Thanks Anthony Drozdek

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[Kamailio-Users] DTLS with kamailio
by mustafa samara 31 Jan '10

31 Jan '10

Hello I want to try DTLS with kamailio, I am working on minisip it support DTLS, but what a bout the kamailio side? is there any thing to do? also i would ask about IPSec support is there any thing to do on kamailio side, or it just on the kernel level? any help please? thanks -- Eng.Mustafa Al-Samara

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[Kamailio-Users] tcp_blocking_connect related errors and Kamailio using 100% CPU
by Pascal Maugeri 29 Jan '10

29 Jan '10

Hi Using Kamailio 1.5 we periodically get the following errors: Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (110) Connection timed out Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_send: connect failed Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:tm:msg_send: tcp_send failed They are several emails in this mailing list explaining we should play with tcp parameters. I will try to change my configuration and see if we can solve this issue. But when this kind of error happen we observe that the kamailio processes are using 100% of CPU (top). Our client is using SIP over TCP only (no UDP). Do you have any hint on how to configure Kamailio to improve TCP support ? Is it normal to kamailio processes to overload CPU ? Regards, Pascal

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Re: [SR-Users] TLS problems
by Andreas Rehbein 29 Jan '10

29 Jan '10

Hi, I solved the problem within snom370 registration with TLS: It was necessary to set the TLS-Modul Parameter "cipher_list" to RSA. Now the snom 370 registers via TLS. Thanks Klaus! regards Andreas

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Re: [SR-Users] TLS problems
by Andreas Rehbein 29 Jan '10

29 Jan '10

Hello Klaus, thank you very much! I really appreciate your help! I followed your instructions and I hope that the attached file includes the necessary informations. regards Andreas -----Ursprüngliche Nachricht----- Von: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Gesendet: Freitag, 22. Januar 2010 10:50 An: Andreas Rehbein Betreff: Re: AW: [SR-Users] TLS problems Andreas Rehbein schrieb: > Hi Klaus! > > Yes, the crash happens everytime (and immediately) when I push the > "Register"-Button in snoms web-gui. > > I attached the backtrace-file, but until now I did not rebuild kamailio with > -DDBG_QM_MALLOC. Please let me know if it's necessary. the log only shows the last function call. Please start gdb again and at the gdb prompt generate a backtrace by entering bt, e.g: (gdb) bt regards klaus > > regards > Andreas > > -----Ursprüngliche Nachricht----- > Von: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Gesendet: Freitag, 22. Januar 2010 09:35 > An: Andreas Rehbein > Cc: sr-users(a)lists.sip-router.org > Betreff: Re: [SR-Users] TLS problems > > Hi Andreas! > > Maybe this is the same bug I reported yesterday. Unfortunately I can not > reproduce the crash anymore. > > Does the crash happen everytime? > > Kamailio produced a core dump: > > 0(6257) ALERT: <core> [main.c:725]: core was generated > > So please send a the backtrace: > # gdb kamailio /path/to/corefile > > bt > > Location of the core file is usually / or /tmp or the kamailio's working > directory. Use > # cat /proc/sys/kernel/core_pattern > to find the exact location of the core dump. > > Further, it might be useful to enable memory debugging: > - in Makefile.defs use -DDBG_QM_MALLOC instead of -DF_MALLOC and > rebuild/reinstall kamailio > - set higher loglevels in kamailio.cfg > memlog=3 > memdbg=3 > debug=3 > modparam("tls", "tls_log", 3) > > Note: this excessive logging makes kamailio start/stop real slow and > logfile will increase rapidly (might fill up your hard disk) > > regards > klaus > > > > Andreas Rehbein schrieb: >> Hi, >> >> >> >> Ive installed Kamailio 3.0 on RHEL5 and activated the mysql backend and >> tls support. Everything works fine with Soft User Agent Phoner lite. >> My Problem is: If I try to register a snom 370 Kamailio crashes >> immediatly. What is wrong? Any suggestions? >> >> Debug: >> >> 17(6290) DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: >> 192.168.0.222 >> >> 17(6290) DEBUG: <core> [tcp_main.c:1052]: tcpconn_new: on port 1430, type > 3 >> 17(6290) DEBUG: <core> [tcp_main.c:1351]: tcpconn_add: hashes: >> 978:1690:1400, 1 >> >> 17(6290) DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8222fe0, 31, >> 2, 0xb613a2b8), fd_no=24 >> >> 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, >> 31, -1, 0x0) fd_no=25 called >> >> 17(6290) DEBUG: <core> [tcp_main.c:3627]: tcp: DBG: sendig to child, >> events 1 >> >> 17(6290) DEBUG: <core> [tcp_main.c:3336]: send2child: to tcp child 0 >> 13(6283), 0xb613a2b8 >> >> 13(6283) DEBUG: <core> [tcp_read.c:884]: received n=4 con=0xb613a2b8, fd=8 >> >> 13(6283) DEBUG: tls [tls_server.c:109]: Using TLS domain TLSs<default> >> >> 17(6290) DEBUG: <core> [tcp_main.c:2826]: DBG: handle_tcp_child: dead >> tcp child 0 (pid 6283, no 13) (shutting down?) >> >> 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, >> 24, -1, 0x0) fd_no=24 called >> >> 17(6290) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 4 >> >> 17(6290) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead >> child 13, pid 6283 (shutting down?) >> >> 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 4, >> -1, 0x0) fd_no=23 called >> >> 0(6257) ALERT: <core> [main.c:722]: child process 6283 exited by a >> signal 11 >> >> 0(6257) ALERT: <core> [main.c:725]: core was generated >> >> 0(6257) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD >> >> 17(6290) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 15(6286) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 16(6289) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 12(6281) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 14(6285) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 10(6278) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 6(6269) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 2(6261) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 11(6280) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 1(6260) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 9(6276) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 8(6272) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 3(6262) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 7(6270) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 5(6267) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 4(6265) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 0(6257) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start >> >> 0(6257) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash > table >> 0(6257) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing > semaphores >> 0(6257) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb >> lists >> >> 0(6257) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done >> >> 0(6257) DEBUG: tls [tls_init.c:621]: tls module final tls destroy >> >> 0(6257) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy >> >> 0(6257) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory >> lock >> >> 0(6257) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD >> >> >> >> tia >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> 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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[Kamailio-Users] kamailio 1.5.4
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

Hello, should we package kamailio 1.5.4? 1.5.3 was in october last year, there were some commits around since then. Next week monday? 1.4.x got out of maintenance period, should we do the 1.4.5 (just release tarballs) to mark that? Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [SR-Users] Request for a special parallel + serial forking scenario
by Klaus Feichtinger 29 Jan '10

29 Jan '10

> > Klaus Feichtinger schrieb: > > > >>> Hi, > >>> > >>> I have a special question regarding a mixture of parallel and serial > >>> forking, operated by Kamailio-3.0.0. > >>> > >>> First of all I will give you a short background information, what the > >>> target will be: > >>> > >>> I have a SIP-server on place A, 3 gateways on place B and 3 gateways > >>> on place C). These gateways are registered on SIP server (A) with > >>> different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, > >>> GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST > >>> be signalled on place B and C in parallel (forking) until the call is > >>> finally established over one of the two involved gateways on place B > >>> or C. When the prime gateway(s) (with the highest Q-value) fail (e.g. > >>> they do not send a provisional response within a timeout or send a > >>> negative response), the next gateway(s) should be addressed (with the > >>> next lower Q-value), a.s.o. > >>> > >>> The configuration works well in case that both gateways on place B > >>> and C fail at the same time (BOTH do not send a response or BOTH send > >>> a negative response or one does not send a response and the other one > >>> sends a negative response). However, in case that only ONE of the two > >>> (always in parallel) addressed gateways fails, it does not work as > >>> expected. > >> > >> What is expected? > >> > >> Do you want to trigger failover to lower priority even if one the > >> gateways is still working fine? You can not do that with sip-router. > >> sr implements parallel forking according to RFC3261, this means it > >> will not generate a failure response as long as one of the branches is > >> still active, maybe sending 200 ok. > >> > > The ideal behaviour would be following: > > - try reaching a gateway on place B and C until one gateway on every > > place (B AND C) sends back a positive (provisional) response - e.g. 180 > > ringing, 183 session progress > > > > - in case that one of the two parallel addressed gateways is reachable > > and sends back a provisional response (= "it is alive"), keep it > > > > - in case that the other one (of the two parallel addressed gateways) > > has a problem (negative response 480 / 404 / 486 ...) or is unreachable > > (= |fr_inv_timer_next| timeout), try the next gateway on this specific > place > > > I'm still not sure if I understand it correct. May examples help: > > the call is forwarded to B1 and C1. B1 does not send back anything, C1 > sends back 100 trying. What is the expected behavior? > - CANCEL C1 and send the call to B2 and C2? > - keep C1 and send another call to B2? > - stay with B1 unreachable and possible C1 answering the call? > The second variant with "keep C1 and send another call to B2" is the expected behaviour. > > - the parallelism is necessary, because calls over these gateways are > > addressed to a usergroup which is split to place B and C behind a > > redundant "non-SIP" PBX system with two active halfs); you can not > > suspect if a user on place B or a user on place C will take the call; > > therefore the call has to be sent to both places. > > Ok. > > > However, the favored behaviour is not standard conform - I know. I guess > > therefore I would need a B2BUA, which is able to establish two > > independent (serially forked) preliminary connections to gateways on > > place B and C and interconnects the successful call to the call > > originator. I fear that no possibility will be available with Kamailio / > > a standard SIP server. > > Maybe you can achieve the wanted behavior by modifying tm module to > CANCEL all branches if a single branch fail (similar to processing of > 6xx error response, which also cancels all branches). > > Depending on the exact needed behavior you might be able to do it with > Asterisk. > > regards > klaus > an alternative solution I think about (but have not yet tested) might be following: - SIP server on place A fork the call to SIP servers on place B and C (both with a fix usrloc contact => parallel forking to two fix destinations) - SIP server on place B fork the call to the gateways on place B serial - SIP server on place C fork the call to the gateways on place C serial The advantage of this solution might be that not one server is responsible for making the mixture of serial AND parallel forking. Each server is responsible for making EITHER parallel (= place A) OR serial (= places B and C) forking. Could this be a practicable alternative? regards Klaus F. -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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[Kamailio-Users] new feature: geoip location in cfg
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

Hello, a new module was uploaded to master branch to lookup ip addresses against a geoip database. The module supports many lookups at the same time and returns the details in pseudo-variables. Therefore a new pv class: $gip(pvc=>key). pvc is the id of the container where to store the results of a geoip search, key is an id for the attribute you want to receive. See details on wiki: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#geoip_module_ps… Readme of the module: http://sip-router.org/docbook/sip-router/branch/master/modules/geoip/geoip.… Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] documentation team
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

Hello, following some discussions on users mailing list, seems we may get a group of people willing to digest documentation and contribute to modules' readme or wiki pages. Hopefully Alex Balashov (who volunteered so far) would have some time to coordinate contributions to git docs, for wiki being easier, as anyone has access. I am willing to help as much I can, but time is always a constraint for a dev. I invite you to join IRC channel #sip-router hosted on freenode.net to plan an synchronize easier. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

please cc the mailing list, otherwise your emails will be ignored. do you have the mysql client installed? Can you connect to the mysql server from terminal with 'mysql' command? Cheers, Daniel On 25.11.2009 19:42 Uhr, YÜKSEL wrote: > Dear Daniel, > > Thanks for your response : Actually I am getting an error below when > I try to create DB with `/usr/local/sbin/kamdbctl create` > > debian:/usr/local/src/kamailio-1.5.0/sip-server# > /usr/local/sbin/kamdbctl create > MySQL password for root: > INFO: test server charset > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 105: mysql: > command not found > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 106: mysql: > command not found > Usage: grep [OPTION]... PATTERN [FILE]... > Try `grep --help' for more information. > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary > operator expected > INFO: creating database openser ... > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 71: mysql: command > not found > ERROR: Creating core database and grant privileges failed! > > I`ve applied the instructor one by one but I didn`t understand whay I > am getting this error. > > Regards, > > Yuksel > > > --- On *Wed, 11/25/09, Daniel-Constantin Mierla /<miconda(a)gmail.com>/* > wrote: > > > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error > To: "lazturk" <lazturk(a)yahoo.com> > Cc: users(a)lists.kamailio.org > Date: Wednesday, November 25, 2009, 12:33 PM > > Hello, > > > On 25.11.2009 17:59 Uhr, lazturk wrote: > > Hello, > > > > How did you solve this problem bcz I am getting a similar error ; > > > > which one? the one related to no database engine specified or the > one related to mysql socket file? > > First: updated the kamctlrc file -- see the comments inside. > > Second: check if mysql server is running and if yes, check the > mysql config to see where the server is listening and where the > mysql client tries to connect to. They must be the same. > > Cheers, > Daniel > > > > > Thanks for your feedback. > > > > > > vivi-8 wrote: > > > >> Hi all > >> I have specify the wanted db type (DBENGINE=MYSQL) in the > >> /usr/local/etc/kamailio/kamctlrc > >> then I using "/usr/local/sbin/kamdbctl create" to create MySQL > database, > >> but I got this error: > >> > >> > >> > >> ERROR: database engine not specified, please setup one in the > config > >> script > >> > >> root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim > >> /usr/local/etc/kamailio/kamctlrc > >> > >> root@acer:/usr/local/src/kamailio-1.5.0/sip-server# > >> /usr/local/sbin/kamdbctl > >> create > >> > >> MySQL password for root: > >> > >> INFO: test server charset > >> > >> ERROR 2002 (HY000): Can't connect to local MySQL server through > socket > >> '/var/run/mysqld/mysqld.sock' (2) > >> > >> ERROR 2002 (HY000): Can't connect to local MySQL server through > socket > >> '/var/run/mysqld/mysqld.sock' (2) > >> > >> Usage: grep [OPTION]... PATTERN [FILE]... > >> > >> Try `grep --help' for more information. > >> > >> /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: > unary > >> operator expected > >> > >> INFO: creating database openser ... > >> > >> ERROR 2002 (HY000): Can't connect to local MySQL server through > socket > >> '/var/run/mysqld/mysqld.sock' (2) > >> > >> ERROR: Creating core database and grant privileges failed! > >> > >> > >> > >> Cheers, > >> > >> vivi > >> > >> > >> _______________________________________________ > >> Kamailio (OpenSER) - Users mailing list > >> Users(a)lists.kamailio.org </mc/compose?to=Users(a)lists.kamailio.org> > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > >> > >> > > > > > > -- Daniel-Constantin Mierla > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla * http://www.asipto.com/

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Request for a special parallel + serial forking scenario
by Klaus Feichtinger 29 Jan '10

29 Jan '10

Hi, I have a special question regarding a mixture of parallel and serial forking, operated by Kamailio-3.0.0. First of all I will give you a short background information, what the target will be: I have a SIP-server on place A, 3 gateways on place B and 3 gateways on place C). These gateways are registered on SIP server (A) with different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST be signalled on place B and C in parallel (forking) until the call is finally established over one of the two involved gateways on place B or C. When the prime gateway(s) (with the highest Q-value) fail (e.g. they do not send a provisional response within a timeout or send a negative response), the next gateway(s) should be addressed (with the next lower Q-value), a.s.o. The configuration works well in case that both gateways on place B and C fail at the same time (BOTH do not send a response or BOTH send a negative response or one does not send a response and the other one sends a negative response). However, in case that only ONE of the two (always in parallel) addressed gateways fails, it does not work as expected. It does nothing and/or waits for an eventually negative response or the call setup timeout. Configuration is done as described in the README of the TM module. It is a mixture of the main route and a failure_route in combination with the functions t_load_contacts() and t_next_contacts(). Does anybody have an idea how this problem could be solved or any alternative solution for this special requirement of parallel signalisation in any case? Thanks in advance! Regards, Klaus -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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[Kamailio-Users] rtpproxy session is not automatically deleted if INVITE transaction fails?
by Iñaki Baz Castillo 28 Jan '10

28 Jan '10

Hi, I'm realizing that rtpproxy session is not automatically terminated when an outgoing INVITE transaction fails ([3456]XX response received from downstream), so the rtpproxy session dies after a while due to time expiration. So I've added the following code to my on_reply_route: ------------ onreply_route[X] { if is_method("INVITE") && status =~ "[3456][0-9][0-9]" unforce_rtp_proxy(); ------------ It seems to do the job. However I expected this to be done automatically by rtpproxy module when an INVITE transaction for which a rtpproxy session has been created fails. Do I miss something? or is it the expected behavior? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] new bunch of PVs
by Daniel-Constantin Mierla 28 Jan '10

28 Jan '10

Hello, I added a bunch of new variables to deal easier with next hop address and be able to retrieve micro-seconds in the config file. Check wiki at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#timeval http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#next_hop_address Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Problems with snmpstats in Kamailio-3.0.0
by Santiago Gimeno 28 Jan '10

28 Jan '10

Hi, I'm trying to migrate from K-1.5 to K-3.0.0 and I'm having problems loading the snmpstats module in kamailio-3.0.0. The error is: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: module_loaded By checking the code it looks like module_loaded function is not implemented in sr_module.c anymore. Could this be the problem? Thanks. Best regards, Santi

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Re: [Kamailio-Users] Bad URI
by Jon Farmer 28 Jan '10

28 Jan '10

On 28 January 2010 13:41, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > > On 1/28/10 2:36 PM, Jon Farmer wrote: >>> >>> is the call going via route[2]? >>> >>> Try to print the whole sip message $mb before and after executing your >>> php >>> script. >>> >> >> Yes route[2] >> >> Here is a link to the $mb before and after http://pastebin.com/d4b7ca03 >> > > k, it is good, however $mb is always received message so it is the same > before and after -- just wanted to check if got corrupted. > > I want to see now the r-uri before and after: $ru > > Daniel >> Aha found it... my PHP is returning error when URI has ";user=phone". Daniel you are a star thank you for your help and patience on this one. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

1 0

[Kamailio-Users] Request for a special parallel serial forking scenario
by Klaus Feichtinger 28 Jan '10

28 Jan '10

Hi Kamailio users, I have a special question regarding a mixture of serial and parallel forking, operated by Kamailio-3.0.0. First of all I will give you a short background information, what the target will be: I have a SIP-server on place A, 3 gateways on place B and 3 gateways on place C). All (6) gateways are registered with the same username on SIP server (A), but with different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST be signalled on gateway(s) of place B and C in parallel (forking) until the call is finally established over one of the two involved gateways on place B or C. When (one of) the prime gateway(s) (with the highest Q-value) fail (e.g. they do not send a provisional response within a timeout or send a negative response), the next gateway(s) should be addressed (with the next lower Q-value), a.s.o. The parallelism for signalisation on place B and C is necessary, because of redundancy / safety reasons. Configuration for serial forking is done as described in the README of the TM module. It is a mixture of the main route and a failure_route in combination with the functions t_load_contacts() and t_next_contacts(). The loaded contacts for serial forking differ (from standard scenarios) in that way, that each sequential target consists of a pair of "users" instead of a single user. This configuration works well in case that both gateways on place B and C fail at the same time (BOTH do not send a response or BOTH send a negative response or one does not send a response and the other one sends a negative response). However, in case that only ONE of the two (always in parallel) addressed gateways fails, it does not work as expected. It does nothing and/or waits for an eventually negative response or the call setup timeout. Only then the next gateway-pair will be addressed. => is the behaviour of TM correct/"as designed" to make an AND-relation with both branches and access the failure_route only then, when both branches fail? Does anybody have an idea how this problem could be solved or any alternative solution for this special requirement of parallel signalisation in any case? Thanks in advance! Regards, Klaus -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

1 0

[Kamailio-Users] TM module
by Denis Putyato 28 Jan '10

28 Jan '10

Hello everybody! ERROR: tm [tm_load.c:47]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Kamailio 3.0 was installed from GIT. Kamailio starts success. But the error appears. I think because of this I have a problem with making a call through kamailio. Calls were failed because kamailio send CANCEL or 487 code immediately after send INVITE to some outgoing gateways (after t_relay() has been made). Is this really need load all modules for tm working properly? I read documentation and fount that almost all modules that need tm loaded do not need in my config. Thank you for any help.

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Re: [Kamailio-Users] Bad URI
by Jon Farmer 28 Jan '10

28 Jan '10

2010/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>: > El Miércoles, 27 de Enero de 2010, escribió: >> 2010/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>: >> > El Miércoles, 27 de Enero de 2010, Jon Farmer escribió: >> >> Hi >> >> >> >> Does anyone know why I am getting >> >> >> >> ERROR: parse_uri: bad host in uri (error at char >> >> >> >> from this ( i hidden the domain and IP) >> > >> > How does the domain look? To show it replace the letters but keep the >> > symbols (as underscore and so). >> >> Hi, >> >> Sorry I am not entirely sure what you mean could you give me an example >> please. >> > > Hi, please keep the thread in the maillist. > > I just mean that, to keep your domain hidden, just change the letters but no > the symbols. Example: > > - Real domain: mycompany-cool-voip.com > - Showed domain: lalalaala-lolo-lili.com > > -- > Iñaki Baz Castillo <ibc(a)aliax.net> > Sorry about the wrong reply address, blame gmail :-) Here you go INVITE sip:01684299007@mydomain.net:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.20.240:5060;branch=z9hG4bK-d8754z-ed473d0ab724045f-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:500@78.32.132.193:5060>. To: <sip:01684299007@mydomain.net:5060>. From: "Michele "<sip:500@mydomain.net:5060>;tag=fd5b583a. Call-ID: NzU1MTZiMjZjZGM0ZTY5ZmFhMWRjNDlkNGQxZTZmODM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. User-Agent: 3CXPhoneSystem 7.0.4708.0. Authorization: Digest username="500",realm="mydomain.net",nonce="4b606dcc81900a95d9962d3c737cc570e624d409",uri="sip:01684299007@mydomain.net:5060",response="1ac466e73c744ed5936179152b5b27d4",algorithm=MD5. Content-Length: 278. . v=0. o=3cxPS 444025798656 5368709121 IN IP4 78.32.132.193. s=3cxPS Audio call. c=IN IP4 78.32.132.193. t=0 0. m=audio 9000 RTP/AVP 0 8 3 101. c=IN IP4 78.32.132.193. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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Re: [Kamailio-Users] ACC on Failure route
by Daniel-Constantin Mierla 28 Jan '10

28 Jan '10

On 1/25/10 2:10 PM, alex pappas wrote: > > What exactly is iirc? common abbreviation in net-slang, helping to save the planet, think green, by sending less bits on net :-): http://www.internetslang.com/IIRC.asp Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] problem with ACC module
by Denis Putyato 28 Jan '10

28 Jan '10

Hello everybody! I have a such problem with ACC module. Calls was accounted but only INVITE and BYE requests has appeared in acc table. ACK is absent. Kamailio 3.0 was installed from GIT. Thank you for any help # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 29) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_url", "xxxxxxxxxxxxxxxxxxxxxxxx") modparam("acc", "db_extra", "src_user=$avp(i:2);src_domain=$si;dst_ouser=$avp(i:1);dst_user=$rU;dst_doma in=$rd") route { sip_trace(); setflag(30); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(29); } if (is_method("ACK")) { xlog("L_NOTICE", "ACK detected"); # I see this line in syslog setflag(1); setflag(29); } route("relay"); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { setflag(1); route("relay"); exit; } else { exit; } } } }

2 1

[Kamailio-Users] Dynamic setting of dispatcher route set
by Alex Balashov 27 Jan '10

27 Jan '10

Unfortunately, ds_select_dst()/ds_select_domain() do not seem to take PVs for their first argument (the route set ID from which to choose). Is there any workaround for this, e.g. when the route set is indeterminate until runtime? It would be very inconvenient to have to work around this by implementing own rollover logic when the dispatcher module is so good... perhaps the possibility of a quick and easy patch to the git trunk? Thanks in advance! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 2

[Kamailio-Users] Bad URI
by Jon Farmer 27 Jan '10

27 Jan '10

Hi Does anyone know why I am getting ERROR: parse_uri: bad host in uri (error at char from this ( i hidden the domain and IP) INVITE sip:xxxxxxxx@mydomain.net:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bKea312c9f5673C67A. From: "200" <sip:200@mydoamin.net>;tag=D6C112A0-D880344D. To: <sip:xxxxxxxx@mydomain.net;user=phone>. CSeq: 2 INVITE. Call-ID: f1f44bbc-58cbed36-2ea0cdab(a)xx.xxx.xxx.xxx. Contact: <sip:200@xx.xxx.xxx.xxx:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067. Supported: 100rel,replace. Allow-Events: talk,hold,conference. Authorization: Digest username="200", realm="mydomain.net", nonce="4b605fcd30e54ca4d6b52ffc9fc6128bc355cf71", uri="sip:xxxxxxxx@mydomain.net:5060;user=phone", response="7c82f9511843dc55aeaed27f468efe39", algorithm=MD5. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 254. . v=0. o=- 1264546283 1264546283 IN IP4 xx.xxx.xxx.xxx. s=Polycom IP Phone. c=IN IP4 xx.xxx.xxx.xxx. t=0 0. a=sendrecv. m=audio 10010 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

2 1

Re: [Kamailio-Users] [sr-dev] Rtpproxy transcoder patch
by Henning Westerholt 27 Jan '10

27 Jan '10

On Tuesday 26 January 2010, Vikram Ragukumar wrote: > I came across a post by Stefan Sayer regarding an rtpproxy transcoder > patch http://lists.iptel.org/pipermail/serdev/2008-July/012794.html. > We are interested in having rtpproxy perform transcoding, both in > software and using DSP array cards, and are currently running rtpproxy > version 1.2.1. > Is this link http://user.cs.tu-berlin.de/~sayer/transcoder/ that is > mentioned on the post our starting point or is there an updated > version ? Hi Vikram, I don't know about this particiular case, so I can't give you more informations in this regards. But i think Stefan read this list as well, if you don't get a reply here perhaps you can contact him direcly by mail, if you have not tried this yet. Cheers, Henning

2 1

[Kamailio-Users] Wrong 1.5 wiki Cookbook page?
by Iñaki Baz Castillo 27 Jan '10

27 Jan '10

Hi, under 1.5 cookbook I see some stuf related to Kamailio 3.0. For example "debug" values are those for SR (if I'm not wrong) and the appearing command to change debug leves uses "sercmd" rather than "kamctl": http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#debug Is it correct? -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] LCR module in Kamailio 3.0
by Daniel-Constantin Mierla 27 Jan '10

27 Jan '10

For reference: http://kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 It helps to upgrade to 3.0.0, feel free to add to the page. In this particular case, kex initializes mi engine ad registers core commands. Still don't see the dr_reload, probably the command is not registered by module's mod init. I will check. Cheers, Daniel On 1/27/10 6:43 AM, Denis Putyato wrote: > > After load kex: > > root@ubuntutest:/kamailio# kamctl fifo which > > shv_get > > shv_set > > sip_trace > > dp_reload > > dp_translate > > lcr_reload > > lcr_gw_dump > > lcr_lcr_dump > > cr_reload_routes > > cr_dump_routes > > cr_replace_host > > cr_deactivate_host > > cr_activate_host > > cr_add_host > > cr_delete_host > > uptime > > version > > pwd > > arg > > which > > kill > > ps > > debug > > get_statistics > > reset_statistics > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 9:00 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > > > On 1/26/10 1:01 PM, Denis Putyato wrote: > > root@ubuntutest:/kamailio# kamctl fifo which > > 500 command 'which' not available > > > do you have the kex module loaded? > > Any fifo command that works for you? > > Cheers, > Daniel > > > root@ubuntutest:/kamailio# > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 2:55 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > > > On 1/26/10 12:35 PM, Denis Putyato wrote: > > I do not use lcr routing in config (only gw table is used). > > ok > > > > Not in kamailio 1.5 nor 3.0. I use carrierroute (in 1.5) and now > drouting (in 3.0). > > By the way. Recently you helped me with drouting. Can I ask one more > question about this module? > > When I try to do "kamctl fifo dr_reload" I receive "500 command > 'dr_reload' not available". > > > do: > > kamctl fifo which > > Should list available MI commands, is dr_reload there? > > Cheers, > Daniel > > > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 12:43 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > > > On 1/26/10 10:38 AM, Denis Putyato wrote: > > Daniel thank you. Now it's working > > > welcome! Be sure you understand the idea behind the instances. Not > being a user of lcr don't know exactly the internal architecture, so > as I understood from discussions on mailing lists, you can have more > lcr instance on same kamailio grouped under same value of lcr_id (many > gws and lcr rules with same lcr_id). > > Cheers, > Daniel > > > > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 12:31 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > Hello, > > On 1/26/10 7:33 AM, Denis Putyato wrote: > > Hello everybody! > > I have a such problem with LCR module > > When I add a new gw in gw table via kamctl addgw or direct via mysql > command, never mind, after doing lcr reload, in memory loaded only gw > with lcr_id = 1. > > All other gateways, with lcr_id > 1, not loaded into memory, so I > cannot use them. > > (After lcr reload I do lcr dump and only gateways whis lcr_id = 1 can > be seen) > > Thank you for any help. > > P.S. lcr table, where lcr_id is present too, is empty. It does not > need for processing in my kamailio. Only gw table is used. > > Kamailio 3.0 was installed from GIT. > > maybe is related to lcr_count parameter: > > http://kamailio.org/docs/modules/3.0.x/modules/lcr.html#id2495949 > > AFAIK, there were some changes in lcr by adding instances, which are > hold in lcr_id (gws to same instance have same id). > > Cheers, > Daniel > > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ > > > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] LCR module in Kamailio 3.0
by Denis Putyato 27 Jan '10

27 Jan '10

Hello everybody! I have a such problem with LCR module When I add a new gw in gw table via kamctl addgw or direct via mysql command, never mind, after doing lcr reload, in memory loaded only gw with lcr_id = 1. All other gateways, with lcr_id > 1, not loaded into memory, so I cannot use them. (After lcr reload I do lcr dump and only gateways whis lcr_id = 1 can be seen) Thank you for any help. P.S. lcr table, where lcr_id is present too, is empty. It does not need for processing in my kamailio. Only gw table is used. Kamailio 3.0 was installed from GIT.

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[Kamailio-Users] Rtpproxy transcoder patch
by Vikram Ragukumar 26 Jan '10

26 Jan '10

Hello, I came across a post by Stefan Sayer regarding an rtpproxy transcoder patch http://lists.iptel.org/pipermail/serdev/2008-July/012794.html. We are interested in having rtpproxy perform transcoding, both in software and using DSP array cards, and are currently running rtpproxy version 1.2.1. Is this link http://user.cs.tu-berlin.de/~sayer/transcoder/ that is mentioned on the post our starting point or is there an updated version ? Thanks and Regards, Vikram.

1 0

[Kamailio-Users] core:proxy.c problem
by marius zbihlei 26 Jan '10

26 Jan '10

Hello, There seems to be a design problem in proxy.c mk_proxy(str* name, unsigned short port, unsigned short proto, int is_sips) function. The purpose of this function is to create a proxy and return a pointer to the created structure. The issue arises from the fact that the name (type struct str) member of the proxy structure is not deep copied from the given parameters(refer to the str* name ) (the hostent structure is instead deep copied). This isn't a problem for now but I have worked on a small patch that caches proxies (using add_proxy() and find_proxy()) and ,because of this shallow copy, things are broken. I said that this is a design problem because we can let the shallow copy happen (performance is better), and when needed the caller should provide a buffer that doesn't change (let him do the copy instead). This is not clearly documented but done from /modules(_k)/utils/conf.c. But it this way we may have a memory leak when the proxy is deallocated, because I doubt that the caller keeps track of the allocated buffers(the code in proxy.c doesn't take ownership of the given pointer). This affects functions mk_proxy and mk_shm_proxy in both kamailio(1.5 to speak of) and sip-router. I have created a patch that also does a deep copy of the name, thus removing the need for the caller to bother about the lifetime of the name buffer. Any ideas?! Cheers Marius

2 3

Re: [Kamailio-Users] ACC on Failure route
by alex pappas 26 Jan '10

26 Jan '10

> Hi Daniel, > > This is my ACC configuration: > > # ----- acc params ----- > /* what sepcial events should be accounted ? */ > modparam("acc", "early_media", 1) > modparam("acc", "report_ack", 1) > modparam("acc", "report_cancels", 1) > /* 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", "failed_transaction_flag", 3)* > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > > "src_user=$fU;src_domain=$fd;dst_to_user=$tU;dst_user=$rU;dst_domain=$rd;diversion_uri=$di;destination_uri=$du;display_name=$fn;orig_src_domain=$od;orig_req_uri=$ou;orig_username=$oU;prefered_identity=$pd;display_name_pref_ident=$pn;proxy_ip=$Ri;proxy_port=$Rp;src_ip=$si;dst_dom_uri=$td;user_agent=$ua") > /* uncomment the following lines to enable DB accounting also */ > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "db_url", > "mysql://openser:openserrw@localhost/openser10") > modparam("acc", "db_extra", > > "src_user=$fU;src_domain=$fd;dst_to_user=$tU;dst_user=$rU;dst_domain=$rd;diversion_uri=$di;destination_uri=$du;display_name=$fn;orig_src_domain=$od;orig_req_uri=$ou;orig_username=$oU;prefered_identity=$pd;display_name_pref_ident=$pn;proxy_ip=$Ri;proxy_port=$Rp;src_ip=$si;dst_dom_uri=$td;user_agent=$ua") > > I run kamailio-1.5.3 no TLS > > What exactly is iirc? > > When i call from an user to an other and the call is passing in case of > failure from failure route1 then I get an extra cdr from the failure route1 > BUT when I call pstn and the call goes to Failure route2 then I don't get > the extra CDRs. One important diference between Failure route1 and 2 is that > in the Failure route2 I have an extra *append_branch();* > > Failure Route 1 (default) > > failure_route[1] { > xlog("alx ------- Failure Route 1 -------"); > #n# if (is_method("INVITE") > #n# && (isbflagset(6) || isflagset(5))) { > #n# unforce_rtp_proxy(); > #n# } > > if (t_was_cancelled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > > # uncomment the following lines if you want to redirect the failed > # calls to a different new destination > ##if (t_check_status("486|408")) { > ## sethostport("192.168.2.100:5060"); > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > > Failure Route2 > > > failure_route[2] { > > if(t_was_cancelled()) { > exit; > } > > if(t_check_status("4[0-9][0-9]|5[0-9][0-9]")) > { > > #xlog("ACCOUNTING: > src_user=$fU;src_domain=$fd;dst_to_user=$tU;dst_user=$rU;dst_domain=$rd;diversion_uri=$di;destination_uri=$du;display_name=$fn;orig_src_domain=$od;orig_req_uri=$ou;orig_username=$oU;prefered_identity=$pd;display_name_pref_ident=$pn;proxy_ip=$Ri;proxy_port=$Rp;src_ip=$si;dst_dom_uri=$td;user_agent=$ua;sip_replay_reson=$rr;sip_replay_status=$T_reply_code"); > > if(ds_next_domain()) > { > xlog("alx ------------------------------------- The final RU is > $rU and the cust_prefix:$avp(s:cust_prefix) --------"); > xlog("alx ------- [ FAILURE ROUTE 2 ] First part of failure with > rd = $rd and prefix = $(ru{uri.param,prefix}) ---"); > # more destinations from same dispatcher group > > # add prefix if set and remove the param from R-URI > if($(ru{uri.param,prefix})!=null) > { > $ru = "sip:" + $avp(s:cust_prefix) + > $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; > } else { > $ru = "sip:" + $avp(s:cust_prefix) + $avp(s:user) + "@" + > $rd; > } > #rewriteport(5065); > #setport("5065"); > t_on_failure("2"); > *append_branch();*##################################################################### This > is different !!!!!!!!!!!!!!!!!! > t_relay(); > exit; > } else { > # go to next dispatcher group > xlog("alx ------- This is the next DST group = $avp(s:dstgrp) > -------"); > > > if($avp(s:dstgrp)!=null) > { > # select destination > if(ds_select_domain("$avp(s:dstgrp)", "4")) > { > xlog("alx ------- [ FAILURE ROUTE 2 ] Second part of > failure with rd = $rd and prefix = $(ru{uri.param,prefix}) ---"); > xlog("alx ------------------------------------- The > final RU is $rU and the cust_prefix:$avp(s:cust_prefix) --------"); > # add prefix if set and remove the param from R-URI > if($(ru{uri.param,prefix})!=null) > { > $ru = "sip:" + $avp(s:cust_prefix) + > $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; > } else { > $ru = "sip:" + $avp(s:cust_prefix) + $avp(s:user) > + "@" + $rd; > } > $avp(s:dstgrp) = null; > xlog("alx ------- This is the next DST group after > NULL = $avp(s:dstgrp) -------"); > xlog("alx ------- The final RURI is $ru ------- "); > t_on_failure("2"); > *append_branch();*##################################################################### > This is different !!!!!!!!!!!!!!!!!! > t_relay(); > exit; > } > } else { > t_reply("444", "No more tries for you!"); > } > } > } > > > > Thanks > Alex > > > > > On Mon, Jan 25, 2010 at 12:39 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hi Alex, >> >> >> On 1/25/10 11:03 AM, alex pappas wrote: >> >> Hi Daniel, >> >> We make a call and the call tries from the Failure Route 5 times new >> gateway (with dispatcher) but WITHOUT success in termination. In every try >> we get back 503 Congestion. In this case how many CDRs we should see in our >> LOG. >> >> We have the modparam("acc", "failed_transaction_flag", 6) and we >> setflag(6) for every INVITE. >> >> iirc, at some point was an option to get all failed legs accounted. I >> don't remember using it so I have to check the sources (you are on 1.5.x, >> right?). >> >> For next version I plan to make acc functions available on rest of the >> routes, should be few updates to take sip method from cseq header and ignore >> r-uri if it is a reply processed, in failure route should work by default, I >> see no reasons now what could be the problem. >> >> Cheers, >> Daniel >> >> >> On Fri, Jan 22, 2010 at 3:33 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 1/22/10 1:08 PM, alex pappas wrote: >>> >>>> Dear Friends, >>>> >>>> In case of a call failure I'm trying from failure route to send the call >>>> through other gateways with Dispatcher module. >>>> I would like to have ACC for all failures that a call can possibly pass. >>>> If the call fails in the 3 first gateways I want them also in my mysql ACC >>>> table. I tried that with AVPs in the syslog and it works fine BUT is there >>>> any way that I can simply setflag(1) for accounting and I can have ACC for >>>> every try? >>>> >>>> I tryied to add in failure route : >>>> >>>> setflag(1); # do accounting ... >>>> setflag(3); # ... even if the transaction fails >>>> >>>> but I'm getting ACC only the last try. >>>> >>>> last failed try? Or what you mean by "the last try"? IIRC, failed >>> transaction flag should be set for each leg if you want all failures. >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla >>> * http://www.asipto.com/ >>> >>> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users@lists.kamailio.orghttp://lists.kamailio.org/cgi-bin/mailman/listinfo/usershttp://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> -- >> Daniel-Constantin Mierla >> * http://www.asipto.com/ >> >> >

3 4

[Kamailio-Users] Manipulate Berkeley DB
by Santiago Soares 26 Jan '10

26 Jan '10

Hello, I'd like to use dispatcher module with Berkeley DB. How can I manipulate the dispatcher table, I mean, insert or delete gateways? kamctl seems not to work: [root@vnsiprtest ~]# kamctl dispatcher addgw 1 '1.2.3.4' 1 'test' /usr/local/sbin/kamctl: line 1458: insert into dispatcher ( setid, destination, flags, description ) VALUES (1,'1.2.3.4',1,'test');: command not found ERROR: dispatcher - SQL Error Thank you. Santiago

2 1

[Kamailio-Users] Proxy caching patch
by marius zbihlei 26 Jan '10

26 Jan '10

Hello, I have worked lately on the patch that tries to mitigate one performance problem in core:action.c file. I am talking about the multiple calls to mk_proxy(), and the creation of temporary proxies for each FORWARD_T action that the daemon executes. I think this behaviour can be improved by caching certain proxies. My idea is this: * If the next_hop matches a name in the alias list (for example we have an alias for example.com and the next_hop is proxy1.example.com), then cache the proxy by calling add_proxy instead of mk_proxy. If the proxy is already cached a pointer is returned, if not it is created and added (the functionality already exists in proxy.c) * For all other hosts a temporary proxy is made(same as now). This whole logic can be turned off(default)/on by a parameter in .cfg. I have run some tests and this can lead to big improvemets (because mk_proxy mean some pkg_malloc - fast but also some DNS resolving - slow). This might benefit some medium-large setups where you have some servers forwarding request to other application servers, etc. The patch is very small and it's already done and finished. I will provide it tomorrow if somebody thinks this idea is good Cheers Marius

1 0

[Kamailio-Users] 30x Redirect / Forward, even questions
by Samuel Muller 26 Jan '10

26 Jan '10

Hello all, I've questions 'bout the redirect/forward with 30x requests. Actually, it does not work, it seems the uac redirect module doesn' t use the new contact for the new invite. simple schema : pstn gw -> kamailio -> linksys -> call forward unconditionnal to another pstn number -> kamailio -> gw pstn the 302 as seen with ngrep tool : #SIP/2.0 302 Moved Temporarily. #To: <sip:B@sip_proxy;user=phone>;tag=91515579fff6b8d1i0. #From: <sip:A@pstn_gw1;user=phone>;tag=16627-UI-00020cf3-54eea92c4. #Call-ID: 16627-RO-00020cf2-4186e9762@pstn_gw1. #CSeq: 81990 INVITE. #Via: SIP/2.0/UDP sip_proxy;branch=z9hG4bK0f0c.22989b87.1. #Via: SIP/2.0/UDP pstn_gw1:5060;rport=5060;received=ip_pstn_gw1;branch=z9hG4bK-25EA-243F3. #Record-Route: <sip:sip_proxy;lr=on;ftag=16627-UI-00020cf3-54eea92c4;did=f65.bcc44f06>. #Contact: <sip:C@sip_proxy>. #Diversion: "B" <sip:B@sip_proxy>;reason=unconditional. #Server: Linksys/SPA942-6.1.3(a). DBG:uac_redirect:shmcontact2dset: adding contact <sip:C@sip_proxy> dest set : Contact: sip:B@ip_B, <sip:C@sip_proxy>;q=0.01 contact $ct: <sip:pstn_gw1:5060> the second pstn gateway received a request to B, and not C (so it creates a loop through pstn ...). in my config : loadmodule "uac.so" modparam("uac", "rr_store_param", "vsf") modparam("uac", "from_restore_mode", "auto") loadmodule "uac_redirect.so" modparam("uac_redirect", "default_filter", "accept") modparam("uac_redirect","bflags", 1) loadmodule "diversion.so" failure_route[2] { setdebug(4); if (t_check_status("301|302")) { route(26) # acc fwd $avp(s:acc_status) = "cfu"; if (!get_redirects("2:1")) { xlog("L_NOTICE", "failed to extract contact info from 30x header"); route(10); # stop rtp t_reply("480", "Temporarily Unavailable"); exit; } # flag 3xx status, open a new branch and back to find the new callee setflag(28); append_branch(); t_on_branch("1"); route(3); # search the new callee setdebug(); exit; } t_on_branch("1"); setdebug(); route(3); # find callee route(12); # local outbound route(10); # stop rtp } I'm using lcr module to route calls, kamailio v. 1.5.3. I've another question, is the 30x always considered as a failure_route ? Any help and/or explanations about the tricks would be really appreciated ... thanks, -- Samuel MULLER sml(a)l33.fr

2 4

[Serusers] Questions about the command : serctl fifo
by inge 26 Jan '10

26 Jan '10

Hi, I am using SER 0.9.9. I have questions about two commands "serctl fifo" : - serctl fifo t_stats : What does mean the "current", "waiting" and "total" values returned by the command? - serctl fifo ul_stats : What does mean the two values returned on the line "location"? Regards, David.

1 0

[Kamailio-Users] Global constants
by Alex Balashov 26 Jan '10

26 Jan '10

Daniel, What is the recommended way (if any) to have a "global constant" that exists outside of any particular routes, in K3? My objective is to define a global DB_URL that can be used by all modules everywhere, but I can't just 'and' the modules together, e.g. modparam("mod1|mod2|mod3|...", "db_url", "...") because the modules are dynamically included (thanks to include_file!) and I do not know the list of all loaded modules easily at runtime. So instead, db_url is configured on a per-module basis. I thought preprocessor constants (e.g. #!define and friends) would allow me to set values for them, but they don't, so that's out. What other options are available that do not involve an event/local route, and operate instead in the completely global namespace? Right now, I am using a Perl script that manually substitutes the value of the db_url modparam everywhere in a large directory tree. :/ What am I not thinking of? Cheers, -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 1

[Kamailio-Users] Active dialogs waaaay to high
by Anders 26 Jan '10

26 Jan '10

Hi, When I run 'kamctl fifo get_statistics all calls', it tells me that the the number of dialogs is about 2000 times higher than what it really is. Actually it seems like it has added a couple of thousand to the number. How do I 'reset' this number? A restart of kamailio didn't help... Thanks! //Anders

2 1

[Kamailio-Users] "Permanent" AVP
by Antonio Reale 26 Jan '10

26 Jan '10

Hi all, I'd like to simulate in kamailio 1.4.4, the loading of a global setting (like a parameter that is loaded only at startup), using AVPs. So, I want to load some AVP only at proxy startup. Probably it is not possible because, if I'm not wrong, AVPs are transaction persistent. The problem is that I don't need to load those AVPs for every transaction. There is a why to implement such configuration? Thanks. Best regards. -- Antonio

3 5

[Kamailio-Users] 477 Send failed
by vragukumar＠signalogic.com 25 Jan '10

25 Jan '10

Hello, I have setup Kamailio and rtpproxy on a server, that acts as a SIP Proxy. An asterisk server sits behind this proxy server. I have noticed that on some occasions when the softphone(x-lite) tries to register with the asterisk through the proxy,it receives a 477 send failed. If i restart Kamailio, the problem seems to disappear. I have attached a logfile showing Kamailio's error output. What could be the reason for such behavior ? Thanks and Regards, Vikram.

3 2

[Serusers] how to compile the docs?
by Chump Chumpster 25 Jan '10

25 Jan '10

Hi, I'm sure this is a simple question and hopefully a simple answer. I've downloaded and am experimenting with SER 2.0.x. I can't find any instructions on how to compile the docs included with the source, into a usable format. They're pretty difficult to use as XML without a stylesheet :-) I appreciate any advice to get me going in the right direction....

1 0

[Kamailio-Users] Fwd: TLS Multiple records not read
by Jijo Jose 25 Jan '10

25 Jan '10

Thanks for the response. The SIP server is reading only the first record(first sip message), the _tls_read() function's log shows the length of first sip message which is matching with length of first record in wireshark.. This means SSL_read returns only the first record. I did the test using TCP which is working fine as it is reading two sip messages on a single read. Please find the log and attached print screen of wireshark traces. Jan 22 20:41:37 REGS-1a0240 openser[936]: io_wait_loop_sigio_rt: siginfo: signal=35 (35), si_code=1, si_band=0x41, si_fd=37 Jan 22 20:41:37 REGS-1a0240 openser[936]: TCPCONN: handle_io: fd map 0x1011d73c (37): {37, 2, 0x30294da0} Jan 22 20:41:37 REGS-1a0240 openser[936]: tls_update_fd: New fd is 37 Jan 22 20:41:37 REGS-1a0240 openser[936]: _tls_read: 559 bytes read Jan 22 20:41:37 REGS-1a0240 openser[936]: read= 559 bytes, parsed=559, state=4, error=1 Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: last char=0x0A, parsed msg= BYE sip:551138175007@21.21.81.11:6061;transport=tls SIP/2.0^M To: SIP5007_TLS_SIMPLEX <sip:551138175007@41.41.0.230<sip%3A551138175007(a)41.41.0.230>>;tag=91c2c894c0^M From: <sip:551138175008@41.41.0.230:5061;transport=tls>;tag=snl_G84KdX4MLT^M Call-ID: 0ad3e453326a4160^M CSeq: 1 BYE^M Route: <sip:21.21.27.10:5061;transport=tls;ftag=91c2c894c0;lr=on>^M Via: SIP/2.0/TLS 41.41.0.230:5061;branch=z9hG4bK_brancha_41.41.0.230_KonTU5DMPx^M Accept-Language: en;q=0.0^M Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER^M Date: Fri, 22 Jan 2010 20:41:36 GMT ^M Max-Forwards: 69^M Content-Length: 0^M ^M Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: end of header part Jan 22 20:41:37 REGS-1a0240 openser[936]: - received from: port 5061 Jan 22 20:41:37 REGS-1a0240 openser[936]: - received from: ip 41.41.0.230 Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: headers: BYE sip:551138175007@21.21.81.11:6061;transport=tls SIP/2.0^M To: SIP5007_TLS_SIMPLEX <sip:551138175007@41.41.0.230<sip%3A551138175007(a)41.41.0.230>>;tag=91c2c894c0^M From: <sip:551138175008@41.41.0.230:5061;transport=tls>;tag=snl_G84KdX4MLT^M Call-ID: 0ad3e453326a4160^M CSeq: 1 BYE^M Route: <sip:21.21.27.10:5061;transport=tls;ftag=91c2c894c0;lr=on>^M Via: SIP/2.0/TLS 41.41.0.230:5061;branch=z9hG4bK_brancha_41.41.0.230_KonTU5DMPx^M Accept-Language: en;q=0.0^M Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER^M Date: Fri, 22 Jan 2010 20:41:36 GMT ^M Max-Forwards: 69^M Content-Length: 0^M ^M . Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: content-length= 0 Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: body: Jan 22 20:41:37 REGS-1a0240 openser[936]: calling receive_msg(0x30294e14, 559, ) Jan 22 20:41:37 REGS-1a0240 openser[936]: SIP Request: Jan 22 20:41:37 REGS-1a0240 openser[936]: method: <BYE> Jan 22 20:41:37 REGS-1a0240 openser[936]: uri: <sip:551138175007@21.21.81.11:6061;transport=tls> Jan 22 20:41:37 REGS-1a0240 openser[936]: version: <SIP/2.0> Thanks Jijo On Sun, Jan 24, 2010 at 5:53 AM, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > I would verify if the received fragment really contains 2 complete SIP > messages (e.g. 2xCRLF at the end of the SIP headers and if Content-Length > header is correct). > > For debugging, TLS is PITA. > > Maybe the SIP server shows the same behavior when using TCP. Otherwise you > can try to configure the NULL:CIPHER on both servers - then you see the > plaintext SIP message in the TLS packets. > > regards > klaus > > Jijo Jose wrote: > >> Hi All, >> We have a SIP server which enabled NAGLE algorithm and proxy as openser >> ver 1.1 >> SIP Server send a TLS multiple records( 2 SIP messages) in a packet to >> openser. Openser is reading only the first record( first SIP message). The >> second TLS record(sip message) is read only when the next message recieved >> by openser from the SIP Server. >> I would like to know anybody observed this issue. I have compared the >> codebase of kamailio 1.4 and openser 1.1 but didin't find any diffrence for >> TLS. >> I have looked at the API _tls_read() in openser which is same as that of >> kamilio. >> Don't we need to use SSL_pending() after SSL_read() to verify any buffer >> is left in the SSL layer ? >> Do you think is it due to the poll method error? The poll method we use >> is POLL_SIGIO_RT >> Please let me know your comments. >> Thanks in advance. >> Jijo >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > >

2 4

[Kamailio-Users] load drouting
by Denis Putyato 25 Jan '10

25 Jan '10

Hello everybody! Such error happens when I try start kamailio Jan 25 10:52:57 ubuntutest kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/drouting.so>: /usr/local/lib/kamailio/modules_k/drouting.so: undefined symbol: db_bind_mod Jan 25 10:52:57 ubuntutest kamailio: : <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 42, column 12-24: failed to load module ls /usr/local/lib/kamailio/modules_k/ acc.so db_text.so group.so nathelper.so pv.so sl.so uac_redirect.so alias_db.so dialog.so htable.so nat_traversal.so qos.so sms.so uac.so auth_db.so dispatcher.so imc.so path.so ratelimit.so speeddial.so uri_db.so auth_diameter.so diversion.so kex.so pdt.so registrar.so sqlops.so userblacklist.so auth.so domainpolicy.so maxfwd.so permissions.so rr.so sst.so usrloc.so benchmark.so domain.so mi_datagram.so pike.so rtimer.so statistics.so xlog.so call_control.so drouting.so mi_fifo.so presence_mwi.so siptrace.so textops.so cfgutils.so exec.so msilo.so pua_mi.so siputils.so tmx.so In config file: loadpath "/usr/local/lib/kamailio/modules:/usr/local/lib/kamailio/modules_k" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "siptrace.so" loadmodule "dialplan.so" loadmodule "lcr.so" loadmodule "sqlops.so" loadmodule "carrierroute.so" loadmodule "drouting.so" Kamailio ver 3.0 was installed from GIT. Thank you for any help

2 4

[Kamailio-Users] Outgoing registrations
by Andrey Utkin 25 Jan '10

25 Jan '10

How do you make outgoing registrations to sip providers' accounts? I haven't found Kamailio functionality like this, so looks this requires some b2bua usage? Is there some other way?

3 6

[Kamailio-Users] Option ac_extra_hdrs to tm in K 3.0.x
by Alex Balashov 25 Jan '10

25 Jan '10

What exactly is the appropriate syntax for the value of the ac_extra_hdrs option in K3's 'tm'? As described here, it is not very clear: http://www.kamailio.org/docs/modules/3.0.x/modules/tm.html#ac_extra_hdrs modparam("tm", "ac_extra_hdrs", "myfavoriteheaders-") What does this mean? Where do I put the header name, and what is the delimeter? Is it a semicolon or colon-separated list? What if, for example, I want to pass X-Alex-Status and X-Alex-Cause in CANCELs and negative ACKs from the original INVITE? Do I use ... modparam("tm", "ac_extra_hdrs", "X-Alex_Status;X-Alex-Cause") If so, what is the significance of the '-' in the above example? Thanks! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 3

[Kamailio-Users] ACC on Failure route
by alex pappas 25 Jan '10

25 Jan '10

Dear Friends, In case of a call failure I'm trying from failure route to send the call through other gateways with Dispatcher module. I would like to have ACC for all failures that a call can possibly pass. If the call fails in the 3 first gateways I want them also in my mysql ACC table. I tried that with AVPs in the syslog and it works fine BUT is there any way that I can simply setflag(1) for accounting and I can have ACC for every try? I tryied to add in failure route : setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails but I'm getting ACC only the last try. Thanks Alex

3 11

[Kamailio-Users] TLS Multiple records not read
by Jijo Jose 24 Jan '10

24 Jan '10

Hi All, We have a SIP server which enabled NAGLE algorithm and proxy as openser ver 1.1 SIP Server send a TLS multiple records( 2 SIP messages) in a packet to openser. Openser is reading only the first record( first SIP message). The second TLS record(sip message) is read only when the next message recieved by openser from the SIP Server. I would like to know anybody observed this issue. I have compared the codebase of kamailio 1.4 and openser 1.1 but didin't find any diffrence for TLS. I have looked at the API _tls_read() in openser which is same as that of kamilio. Don't we need to use SSL_pending() after SSL_read() to verify any buffer is left in the SSL layer ? Do you think is it due to the poll method error? The poll method we use is POLL_SIGIO_RT Please let me know your comments. Thanks in advance. Jijo

2 1

[Kamailio-Users] state field on dialog table
by Panagiotis Skoulikaritis 23 Jan '10

23 Jan '10

Hello I have a question about the state field on the dialog table. I guess it is more sip protocol oriented than kamailio deployment, I assume that the values (1-5) that are logged are the state of the dialog correct ? and if the above are correct the dialog states are the following ? 1. waiting for UAC deciscion 2. Initiate dialog 3. Establish dialog 4. Stop Media 5. Destroy dialog I apologize for my sip ignorance Trying to build up my knowledge. :) I' m asking this because I' m trying to export statistics about the calls that are handled from the kamailio and the state of the call for specific users,IP's, destinations. Thanks Panagiotis

4 5

[Kamailio-Users] mysql loadbalancing
by BERGANZ François 22 Jan '10

22 Jan '10

Hello, I need to know if Kamailio can change Mysql Server if there is a problem with a mysql server ? Can Kamailio do loadbalancing between 2 mysql servers which are duplicate ? Thank you François BERGANZ

5 17

TLS problems
by Andreas Rehbein 22 Jan '10

22 Jan '10

Hi, I've installed Kamailio 3.0 on RHEL5 and activated the mysql backend and tls support. Everything works fine with Soft User Agent "Phoner lite". My Problem is: If I try to register a "snom 370" Kamailio crashes immediatly. What is wrong? Any suggestions? Debug: 17(6290) DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: 192.168.0.222 17(6290) DEBUG: <core> [tcp_main.c:1052]: tcpconn_new: on port 1430, type 3 17(6290) DEBUG: <core> [tcp_main.c:1351]: tcpconn_add: hashes: 978:1690:1400, 1 17(6290) DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8222fe0, 31, 2, 0xb613a2b8), fd_no=24 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 31, -1, 0x0) fd_no=25 called 17(6290) DEBUG: <core> [tcp_main.c:3627]: tcp: DBG: sendig to child, events 1 17(6290) DEBUG: <core> [tcp_main.c:3336]: send2child: to tcp child 0 13(6283), 0xb613a2b8 13(6283) DEBUG: <core> [tcp_read.c:884]: received n=4 con=0xb613a2b8, fd=8 13(6283) DEBUG: tls [tls_server.c:109]: Using TLS domain TLSs<default> 17(6290) DEBUG: <core> [tcp_main.c:2826]: DBG: handle_tcp_child: dead tcp child 0 (pid 6283, no 13) (shutting down?) 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 24, -1, 0x0) fd_no=24 called 17(6290) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 4 17(6290) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead child 13, pid 6283 (shutting down?) 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 4, -1, 0x0) fd_no=23 called 0(6257) ALERT: <core> [main.c:722]: child process 6283 exited by a signal 11 0(6257) ALERT: <core> [main.c:725]: core was generated 0(6257) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD 17(6290) INFO: <core> [main.c:788]: INFO: signal 15 received 15(6286) INFO: <core> [main.c:788]: INFO: signal 15 received 16(6289) INFO: <core> [main.c:788]: INFO: signal 15 received 12(6281) INFO: <core> [main.c:788]: INFO: signal 15 received 14(6285) INFO: <core> [main.c:788]: INFO: signal 15 received 10(6278) INFO: <core> [main.c:788]: INFO: signal 15 received 6(6269) INFO: <core> [main.c:788]: INFO: signal 15 received 2(6261) INFO: <core> [main.c:788]: INFO: signal 15 received 11(6280) INFO: <core> [main.c:788]: INFO: signal 15 received 1(6260) INFO: <core> [main.c:788]: INFO: signal 15 received 9(6276) INFO: <core> [main.c:788]: INFO: signal 15 received 8(6272) INFO: <core> [main.c:788]: INFO: signal 15 received 3(6262) INFO: <core> [main.c:788]: INFO: signal 15 received 7(6270) INFO: <core> [main.c:788]: INFO: signal 15 received 5(6267) INFO: <core> [main.c:788]: INFO: signal 15 received 4(6265) INFO: <core> [main.c:788]: INFO: signal 15 received 0(6257) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(6257) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(6257) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(6257) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(6257) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(6257) DEBUG: tls [tls_init.c:621]: tls module final tls destroy 0(6257) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(6257) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock 0(6257) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD tia

2 4

[Kamailio-Users] Asterisk 403 Forbidden error with port translation
by Vikram Ragukumar 21 Jan '10

21 Jan '10

Hello, I have made some progress since my previous post, but not enough :). ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the outbound proxy for the softphone(Outbound proxy a.b.c.d:9090). Authentication credentials for the softphone match the user registered in asterisk's sip.conf. F.W runs Kamailio and rtpproxy, with Kamailio listening on port 5060. The asterisk server is setup to listen on port 5060. The Firewall(F.W), uses a libnetfilter_queue based program to : (a) Rewrite the destination port 9090 as 5060, and rewrite all other occurrences of 9090 as 5060 in the SIP message, for packets from the softphone to the asterisk server. (b) Rewrite the source port 5060 as 9090, and rewrite all other occurrences of 5060 as 9090 in the SIP message, for packets from the asterisk server to the softphone. The following exchange of SIP messages take place -Sip softphone sends a REGISTER message to asterisk -Asterisk responds with a 401 UNAUTHORIZED -Sip softphone replies with a REGISTER message containing auth. info. -Asterisk responds with a 403 FORBIDDEN : BAD AUTHORIZATION The above setup works when the softphone uses port 5060, so there problem here does not have anything to do with Authorization credentials. Is it possible i might be modifying parts of the packet that shouldn't be modified or i might not be modifying some relevant parts of the packet ? Thanks in advance, Vikram.

3 3

[Serusers] Hello , Help, and Thanks
by Justin 21 Jan '10

21 Jan '10

Hi, I have just begun to set up my soft phone, and am currently reading through your How to's Ser text's ,,, but I keep getting error 401-403-404-408 all doing with I am entering the wrong DOMAIN and PROXY for the Iptel.org SIP service. If you could contact me with a solution. Much Thanks (dark4ajaguar(a)iptel.org)

1 0

[Kamailio-Users] Script variables Question
by alex pappas 21 Jan '10

21 Jan '10

Dear friends, When a sip 4rU is entering my kamailio script I'm cutting down the prefix of this customer and added to a script variable. In the same same route after i do some checks then i add back the customer prefix which I had kept in the script variable. The problem: Some times the call goes to failure Route for rerouting and then i try to add the prefix again but the is empty. How can I transfer a variable from route to route and be a unique variable for each call? Thank you Alex

5 5

[Kamailio-Users] concurrent dialogs
by alex pappas 21 Jan '10

21 Jan '10

Dear Friends, I have the following question: When I'm running the command *kamctl fifo dlg_list* I'm getting the concurrent dialogs in that moment. When I disable the sip_trace module(which I don't see how they are related) I get NO output when I run the same command. Does anyone knows on why this is happening ? Alex

2 2

[Kamailio-Users] kamailio and sip-router at FOSDEM 2010 - social event
by Henning Westerholt 21 Jan '10

21 Jan '10

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2010/) takes place in Brussels. As in the last year we would like to meet here for a social event, probably a dinner on Saturday evening, 6th February. This developers already confirmed their participance: - Daniel-Constantin Mierla - he is also giving a talk about Kamailio 3.0.0 on Saturday: http://fosdem.org/2010/schedule/events/kamailio_sip_server - Elena-Ramona Modroiu - Marius Zbihlei - 1&1 kamailio and sip-router development Bucharest - Timo Reimann - 1&1 kamailio and sip-router development Karlsruhe - Henning Westerholt Participation at the event is free of charge, everybody pays for its own expenses. In order to organize a restaurant and do a reservation please send me a short mail if you would like to attend. Suggestions for a nice place with good food and beer are also welcome. Looking forward to see you in Brussels. Best regards, Henning P.S.: Photos of the last year event can be found on our web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 6

[Kamailio-Users] Question about Timer, i need milliseconds
by Marco Bungalski 20 Jan '10

20 Jan '10

Hello all :-) I have a question about how to get a time-stamps with milliseconds. To create my own CDR's, i write to my databases timestamps, with a command like this: sql_query("ca", "UPDATE active_calls SET state = 'BILLING', end_time=$Ts WHERE call_id='$ci'", "ra"); I use the "$Ts"-variable, but with this i can get only "full" seconds. But i need milliseconds for my billing-system. Otherwise i have differences between my CDR's and my carrier's one. Example: A call has a duration of 0.2 seconds. In my CDR start- and endtime is the same (=0 seconds), to my carrier a have 1 second to pay ;-) Is there a possibility to get a time-stamp or time with milliseconds? Cheers, Marco Marco Bungalski GmbH Traversale 5 27283 Verden Telefon: +49 4231 - 776 9999 Fax: +49 4231 - 776 9998 Mobil: +49 172 4204774 e-mail: Marco(a)Bungalski.de web: www.bungalski.de Geschäftsführender Gesellschafter: Marco Bungalski Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586

3 2

[Kamailio-Users] 3G+ and LTE forum - meeting in Berlin
by Daniel-Constantin Mierla 20 Jan '10

20 Jan '10

Hello, I will be in Berlin for few days (Jan 20-22), attending 3G+ and LTE Forum. If anyone is around and want to meet, feel free to drop me an email. Next event I will talk about Kamailio and SIP Router is FOSDEM in Brussels -- more details will follow soon. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] BYE packet is not received in siptrace module !!
by Dinesh Gautam 20 Jan '10

20 Jan '10

Dear all, I am using siptrace module and getting all the packets except the BYE. when the call disconnected BYE packet is not received. I am also getting the packets OK, REGISTER, INVITE two times in sip_trace table. Can anyone help me to sort out this problem?? -- Dinesh Gautam !!

4 14

[Serusers] deactivate the Via field
by "Martínez García, Rebeca" 20 Jan '10

20 Jan '10

Hello all, I know it might sound strange but, is there any way to deactivate that SER puts its Via on the SIP message when forwarding it to other entity? Thanks in advance, Rebeca Martinez

1 0

Re: [Kamailio-Users] [kamailio-users] Implement kamailio sever failover
by Klaus Darilion 20 Jan '10

20 Jan '10

ram schrieb: > in that case we are going to loose registration right ? not if you do database replication of "location" table > and ongoing calls will be dropped right ? No. only ongoing transactions may be affected. regards klaus PS: please always Cc the mailing list > > Ram > > > > On Sat, Jan 16, 2010 at 2:58 AM, Klaus Darilion > <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> wrote: > > Hi! > > For failover there are several possibilities. The simplest on, I > guess the one you are referring to, is IP takeover. > > This is: 2 servers. On both is Kamailio and the database (e.g. > mysql) installed. The SIP service uses a virtual IP address which is > either configured at server 1 or server 2 (never on both). Kamailio > process is only running on the server which has the virtual IP > address provisioned. > > Then you need some other software to monitor the SIP service and > trigger the failover from one server to the other server (e.g. > either write some shell scripts or use existing software like > hearbeat, ultramonkey, ...) > > Further - depending on your scenario - you might need some database > replication (e.g. mysql cluster or mysql with master-master > replication). > > regards > klaus > > RAJNIKANT VANZA wrote: > > Hi, > > I want to implement kamailio server failover. > so, anybody know about how to configure kamailio server failover? > > e.g. 2 kamailio severs run and when any one server fail then > other server take over thats resources. > > > -- > Best Regards, > > Rajnikant Vanza > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > >

3 2

[Kamailio-Users] no answer to invite
by Anders 19 Jan '10

19 Jan '10

Hi, I have a problem that I don't understand at all (yet another one ;-)). One client has Asterisk (like many others) from where he sends calls to our Kamailio. It worked fine, until it suddenly stopped for no apparent reason on Friday. No changes in configuration on his or our side. We can ping his server just fine, and vice versa. On any INVITE he sends, he gets no, zero, zip response from Kamailio on our server. I wanted to figure out why, so I put an 'ngrep' on port 5060 for his IP on our Kamailio server, and I got...nothing. No communication at all on this port containing this IP. Why? I have no clue. The ping should show that we can communicate, and a 'ngrep' on his server shows that he sends plenty of INVITEs, but gets absolute zero response from our Kamailio box. Any ideas where to look? Thanks a lot in advance!

3 3

[Kamailio-Users] siremis on sourceforge.net
by Elena-Ramona Modroiu 19 Jan '10

19 Jan '10

Hi, the sources of siremis are in sourceforge.net svn. If any of you has developed (plan to develop) extensions, contact me to grant devel access. Also, patches should be sent against latest svn trunk (no branches so far): http://sourceforge.net/projects/siremis/ The svn repo does not store the openbiz core framework -- you have to get it from siremis-1.0.0.tgz (the openbiz dir) or download it from openbiz project and apply the patch from misc directory (note that you also have to download zend and others according to openbiz guides). Once you fetch siremis svn and add openbiz, run make to create the symlinks required for web dirs and you are set. Regards, Ramona

1 0

[Kamailio-Users] SIREMIS v1.0.0 Released
by Elena-Ramona Modroiu 18 Jan '10

18 Jan '10

Hi, I have just released SIREMIS 1.0.0 - web admin interface for Kamailio (OpenSER) 3.0.0 and SIP Router. Among new features: * presence service management – watchers, active watchers and presentity * sip trace view – access sip_trace module records, highlight headers and group by call-id * user location statistics charts – online user agents, supported SIP methods, contacts and nat related charts * update of the core components to latest stable versions * update to work with latest Kamailio (OpenSER) v3.0.0 * patch to work with older Kamailio (OpenSER) v1.5.x * time-based X-axis for charts print labels as HH:MM * charts types supported: line, line dot and area Some screenshots specific for this version: http://www.asipto.com/gallery/v/siremis/siremis_22.png.html http://www.asipto.com/gallery/v/siremis/siremis_23.png.html http://www.asipto.com/gallery/v/siremis/siremis_24.png.html Download and installation steps: http://siremis.asipto.com/install/ More screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Regards, Ramona

1 0

[Kamailio-Users] setbflag not working?
by Ricardo Martinez 18 Jan '10

18 Jan '10

Hello. I have the next situation with kamailio 3.0.0 In my config file I have something like this : route If INVITE then route(AUTH_REQUEST) . . . route[AUTH_REQUEST] . . . if (client_nat_test("3")) { append_hf("P-hint: route(AUTH_REQUEST)|setflag7,forcerport,fix_contact\r\n"); if (setbflag("7")) { xlog("L_INFO","[$ci]:[AUTH_REQUEST] : setbflag(7)"); } force_rport(); fix_contact(); xlog("L_INFO","[$ci]:[AUTH_REQUEST] : CLIENT_NAT_TEST-3"); }; . . . route(IN_TO_IN) route[IN_TO_IN] . . . route(LOCAL_CALL) . . . route[LOCAL_CALL] . . . if (isbflagset("7") ) { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET begin"); } else { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NO SET begin"); } if ( !lookup("location") ) { if ( avp_check("$avp(s:4)", "eq/s:1/g") ) { if( !avp_pushto("$br", "$avp(s:5)/g") ) { sl_send_reply("403","Forbidden - Forward Number NULL"); exit; }; prefix("222"); route(TO_VOICEMAIL_NODISP); exit; }; if ( avp_check("$avp(s:3)", "eq/s:1/g") ) { xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm - From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL ON\n"); route(TO_VOICEMAIL_NODISP); exit; } else { sl_send_reply("480", "Usuario fuera de Servicio"); xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm - From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL OFF\n"); exit; }; }; if (isbflagset("7") ) { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET end"); } else { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NOT SET end"); } . . . For some reason in the route[LOCAL_CALL] the flagb "7" is changed from "SET" to "NOT SET". According to my debugs this unset is made in the "lookup(location)" part. But I don't understand why. Can someone help me here? Thanks in advance, Ricardo Martinez.-

2 2

[Serusers] SER 0.9.7-pre4 and RPID
by Steven C. Blair 18 Jan '10

18 Jan '10

I'm trying to append an RPID header on my SER 0.9.7-pre4 system. The config has not changed and this use to work in SER 0.9.7-pre3. The header does not get appended and I cannot see why. The use_rpid parameter or the auth_db module does not seem to exist in this release. Is there a different way to append the header? Thanks,Steve

1 0

[Kamailio-Users] automatic packaging building of master
by Henning Westerholt 18 Jan '10

18 Jan '10

Hi, we've updated the automatic packaging building process on the devel machine. This builds now every night from the git master packages for debian lenny, squeeze and sid. Hints about your archive configuration in order to use the packages can be found on http://devel.kamailio.org/ They are build from the control files in pkg/debian, so at the moment some packages with special dependencies are missing, and they are named with 'ser-$name'. The only thing that is modified at the moment is the version number. This packages are of course only for tests and not for production use. Please let me know if there are any problems with the building service. Henning

1 0

[Kamailio-Users] %23 remove from rU
by alex pappas 18 Jan '10

18 Jan '10

Dear Friends, Do you have any idea on how to remove from $rU the %23 when is send in to my network? The scenario is: The customer will send (prefix # number). This is translated in sip in (prefix %23 number). I need to remove this in order to check against PDT module. Thanks Alex

4 12

[Kamailio-Users] Kamailio 3.0 Crash when load with module siptrace.
by Quang Tri 18 Jan '10

18 Jan '10

Hello, When i try with module siptrace.so This is 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 "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dialog.so" #!ifdef WITH_CALLCONTROL loadmodule "call_control.so" #!endif #!ifdef WITH_MEDIAPROXY loadmodule "mediaproxy.so" #!endif #!ifdef WITH_SIPTRACE loadmodule "siptrace.so" #!endif .... and with # ----- sip trace ------- #!ifdef WITH_SIPTRACE modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 2) #!endif and this is error log 8(7163) DEBUG: tm [t_lookup.c:764]: DEBUG: t_lookupOriginalT: searching on hash entry 60285 8(7163) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(7163) DEBUG: tm [t_lookup.c:854]: DEBUG: t_lookupOriginalT: no CANCEL matching found! 8(7163) DEBUG: tm [t_lookup.c:856]: DEBUG: t_lookupOriginalT completed 8(7163) DEBUG: tm [tm.c:952]: lookup_original: t_lookupOriginalT returned: (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [receive.c:280]: receive_msg: cleaning up *17(7186) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 14* 17(7186) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead child 5, pid 7160 (shutting down?) 17(7186) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x81ede80, 14, -1, 0x0) fd_no=22 called 0(7151) ALERT: <core> [main.c:722]: child process 7160 exited by a signal 11 0(7151) ALERT: <core> [main.c:725]: core was generated 0(7151) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD 6(7161) INFO: <core> [main.c:788]: INFO: signal 15 received 7(7162) INFO: <core> [main.c:788]: INFO: signal 15 received 16(7185) INFO: <core> [main.c:788]: INFO: signal 15 received 15(7183) INFO: <core> [main.c:788]: INFO: signal 15 received 8(7163) INFO: <core> [main.c:788]: INFO: signal 15 received 14(7179) INFO: <core> [main.c:788]: INFO: signal 15 received 9(7164) INFO: <core> [main.c:788]: INFO: signal 15 received 10(7165) INFO: <core> [main.c:788]: INFO: signal 15 received 13(7176) INFO: <core> [main.c:788]: INFO: signal 15 received 1(7155) INFO: <core> [main.c:788]: INFO: signal 15 received 17(7186) INFO: <core> [main.c:788]: INFO: signal 15 received 2(7156) INFO: <core> [main.c:788]: INFO: signal 15 received 3(7158) INFO: <core> [main.c:788]: INFO: signal 15 received 4(7159) INFO: <core> [main.c:788]: INFO: signal 15 received 11(7171) INFO: <core> [main.c:788]: INFO: signal 15 received 12(7174) INFO: <core> [main.c:788]: INFO: signal 15 received 0(7151) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(7151) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(7151) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(7151) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(7151) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(7151) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(7151) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock 0(7151) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD How can fix it. Thank you.

3 6

[Kamailio-Users] Start/Stop Kamailion
by juergen＠glowka.de 18 Jan '10

18 Jan '10

Hi Dinesh do as Daniel already told You, or use the start/stop-script kamailion.init in the folder debian/ after unpacking the sourcepacket and change it according to Your system. Copy/rename it to /etc/init.d/kamailion check for errors: /usr/loca/sbin/kamailio -c Juergen (who likes india:)

2 1

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