- sr-users - kamailio.org

Re: [SR-Users] Visiting CeBIT 2011 in Hanover (Carsten Bock)
by Dan-Cristian Bogos 23 Feb '11

23 Feb '11

Daniel, I will be there on Thursday/Friday. Cheers. DanB

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422 Session timer too small
by Mino Haluz 23 Feb '11

23 Feb '11

Hi, I see in my cdrtool rarely 422 Session Timer too small - the reason why the call was not initiated. Do you have some suggestion how to fix this ? Maybe SST module ? What is the default value of this timer, so that I could increase it ? kamailio 3.1.0 (i386/linux) 1e204f Thank you, Mino

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kamailio 3.1: forward("10.0.0.10:5060") not working?
by Min Wang 23 Feb '11

23 Feb '11

HI It seems the forward("host:port") not working in the kamailio 3.1? the simple route like: route { forward("67.154.xx.xx:5080"); } the error is: <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "67.154.xx.xx:5080" Anything missing? the detailed log is: 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <load_tm> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_newtran> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_tcp> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_udp> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_forward_nonack> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_release> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [main.c:2371]: Expect (at least) 18 SER processes in your process list 0(8500) DEBUG: <core> [proxy.c:278]: DEBUG: mk_proxy: doing DNS lookup... 0(8500) DEBUG: <core> [dns_cache.c:567]: dns_hash_find(_sip._udp.67.154.xx.xx:5080(28), 33), h=780 0(8500) DEBUG: <core> [resolve.c:727]: get_record: lookup(_sip._udp.67.154.xx.xx.:5080, 33) failed 0(8500) DEBUG: <core> [dns_cache.c:895]: dns_cache_mk_bad_entry(_sip._udp.67.154.xx.xx:5080, 33, 60, 1) 0(8500) DEBUG: <core> [dns_cache.c:828]: dns_cache_add: adding _sip._udp.67.154.xx.xx:5080(28) 33 (flags=1) at 780 0(8500) DEBUG: <core> [dns_cache.c:567]: dns_hash_find(67.154.xx.xx:5080(18), 1), h=16 0(8500) DEBUG: <core> [resolve.c:727]: get_record: lookup(67.154.xx.xx:5080, 1) failed 0(8500) DEBUG: <core> [dns_cache.c:895]: dns_cache_mk_bad_entry(67.154.xx.xx:5080, 1, 60, 1) 0(8500) DEBUG: <core> [dns_cache.c:828]: dns_cache_add: adding 67.154.xx.xx:5080(18) 1 (flags=1) at 16 0(8500) : <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "67.154.xx.xx:5080" 0(8500) ERROR: <core> [route.c:1161]: fixing failed (code=-478) at cfg:/etc/kamailio/kamailio.cfg:394 ERROR: error -478 while trying to fix configuration 0(8500) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(8500) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(8500) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(8500) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(8500) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(8500) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(8500) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock thx. -- Kind regards / Mit freundlichen Grüßen Min Wang

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Re: [SR-Users] How to get the dialog information via dlg_list command?
by yan wang 23 Feb '11

23 Feb '11

Hi Daniel, Thanks a lot for your quick reply. I recompile and install the Kamailio as the following page described.http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-… I also successfully created two SIP users: "User1" and "User2". Now the registeration is passed for the two users. But when "User1" sends INVITE to "User2" through the Kamailio proxy, it always response with 403 "Not relaying" and not forward the INVITE to "User2". Could you please give me some help on this case? Thanks. BTW, I have export the environment variable SIP_DOMAIN to "mysipserver.com". And please check the attached enclosed config file and trace. Best Regards,Spencer --- On Wed, 2/23/11, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] How to get the dialog information via dlg_list command? To: "yan wang" <rtosvip(a)yahoo.com.cn> Cc: sr-users(a)lists.sip-router.org Date: Wednesday, February 23, 2011, 2:42 PM Hello, the problem is you are using the dialog module coming from SER, not the one coming from Kamailio. The loadpath (or mpath) has to include directorins ending in modules_k and modules. I recommend you use kamailio.cfg as a start to build your config and then add the dialog module. Before compiling, when you are using the sources, you have to do: make FLAVOUR=kamailio cfg You can see more at: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git If you prefer to install deb files (on Debian/Ubuntu), see: http://www.kamailio.org/dokuwiki/doku.php/packages:debs Cheers, Daniel On 2/23/11 7:32 AM, yan wang wrote: Dear Friends, I need your help on the following question: I am using Kamailio 3.1.2. I want to get the dialog information dynamically via the FIFO command "dlg_list" by "kamctl fifo dlg_list". But unfortunately, I could get NOTHING in the output. And the fact is that I have setup 5 pairs SIP calls with the Kamailio SIP proxy. Could anyone show me the clues and how to get the correct dialog information? Thanks. BTW, I attached the config file. Best Regards, Spencer _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] get_profile_size() function help (Alex Balashov)
by 侯旭光 23 Feb '11

23 Feb '11

On 02/21/2011 04:50 AM, ??? wrote: > ALTER:<core> [main.c 722] : child process 29651 exited by sinal 11 > ALTER:<core> [main.c 725] : core was generated > INFO :<core> [main.c 737] : terminating due to SIGCHILD This is a crash. It's a bug. Then how to get the number of dialogs belonging to a profile?

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How to get the dialog information via dlg_list command?
by yan wang 23 Feb '11

23 Feb '11

Dear Friends, I need your help on the following question: I am using Kamailio 3.1.2. I want to get the dialog information dynamically via the FIFO command "dlg_list" by "kamctl fifo dlg_list". But unfortunately, I could get NOTHING in the output. And the fact is that I have setup 5 pairs SIP calls with the Kamailio SIP proxy. Could anyone show me the clues and how to get the correct dialog information? Thanks. BTW, I attached the config file. Best Regards,Spencer

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Re: [SR-Users] Unable to start Siremis
by jack＠4voice.net 22 Feb '11

22 Feb '11

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Unable to start Siremis
by jack＠4voice.net 22 Feb '11

22 Feb '11

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UAC Module: test registration to a Voip provider
by Matteo Campana 22 Feb '11

22 Feb '11

Hi all, we are using the UAC module ( http://www.kamailio.org/docs/modules/stable/modules_k/uac.html#id2910015) to register the proxy to an external DID provider. I know that the module takes care of sending REGISTER on the basis of credentials stored in uacreg table, but my question is: if I add a new row in the uacreg table, kamailio will register the new username after the database update or I need a restart of kamailio (or some kamailio module)? If I edit the row in uacreg table and I call the rpc command *sercmd uac.reg_dump *I see the old values in the output, but if I restart kamailio I see the new values. Tanks in advance, Matteo

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how to forward on busy
by x-kamailio＠sidell.org 22 Feb '11

22 Feb '11

I'm trying to do something very basic, which is to forward a call to a different number on busy. But, the various things I've tried from the examples don't work. Starting with the kamailio.cfg that is included with version 3.1, what code would I add to forward all busy calls to "sip:foo@bar.com", a target that is not necessarily local to the server? Many thanks! -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

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Kamailio/SIP Router in RHEL 5
by Suresh Bhandari 22 Feb '11

22 Feb '11

Hi, I wanted to run Kamailio/SIP Server in RHEL 5 (Being an RHCE, I prefer this :)), but the INSTALL document you provided has no description based on that OS :(. I found that there are no RPMs for this as well. Are you preparing them? or Do you have any plans for that? or you prefer me to change the OS. Thanks and Regards, Suresh.

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Visiting CeBIT 2011 in Hanover
by Daniel-Constantin Mierla 22 Feb '11

22 Feb '11

Hello, I am going to visit CeBIT show this year (first week of March in Hanover, Germany), if anyone else here is there and want to meet for a chat about latest developments in the project and VoIP world, drop me an email, maybe we have some overlapping days and can sit together for a bit. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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HTABLE problem
by Ricardo Martinez 21 Feb '11

21 Feb '11

Hello I’m trying to use “htable” module with the example to block a user with three time s failed password. When a REGISTER arrives I’m getting this error : Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:358]: setting pvar failed Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (322,54-322,81) Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:358]: setting pvar failed Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (328,47-328,49) The lines pointing the error are these : 312 if ( !radius_www_authorize("10.0.0.208") ) 313 { 314 switch ($retcode) { 315 case -1: 316 xlog("L_INFO","authentication failed from radius\n"); 317 if($sht(a=>$au::auth_count) == "null") 318 { 319 $sht(a=>$au::auth_count) = 0; 320 } 321 $sht(a=>$au::auth_count) = $sht(a=>$au::auth_count) + 1; 322 if($sht(a=>$au::auth_count) == 3) 323 { 324 xlog("L_INFO","auth failed 3rd time - src ip: $si\n"); 325 } 326 $sht(a=>$au::last_auth) = $Ts; 327 break; 328 case -5: 329 sl_send_reply("403", "Forbidden"); 330 exit; 331 } 332 www_challenge("10.0.0.208", "1"); 333 exit; 334 } What could be the problem? Thanks in advance. Regards, Ricardo.-

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Kamailio Awards 2010
by Daniel-Constantin Mierla 21 Feb '11

21 Feb '11

Hello, almost a tradition by now, being the 4th edition, I published Kamailio Awards for 2010 - a blog post that tries to summarize the top of the activities in the public space related to Kamailio project that happened during the previous year. You can browse it at: http://asipto.com/u/ka10 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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AVP Error in version 3.1.0
by Jijo 21 Feb '11

21 Feb '11

Hi All, On register we store the contact in an avp variable and do a t_relay(). After t_relay() the $avp variable becomes null. I printed the value before after t_relay() to determine this behavior. This happens only on registration load test around 2000 subcribers with ( 4 REGISTER/sec). This happens only for one subscriber out of 2000 subscribers. I did the similar test with $var and its working fine. Anybody observed similar behavior with avp? This was working in kamailio 1.4 version. We did the upgrade recently to 3.1.0 and started observing this issue. How do we debug this issue.? Thanks Jijo

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get_profile_size() function help
by 侯旭光 21 Feb '11

21 Feb '11

Hello bother again what's the problem with the function get_profile_size() My script modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_mode", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "profiles_no_value", "inbound ; outbound") ... if(is_method("INVITE")) { ... dlg_manage(); set_dlg_profile("inbound"); ... get_profile_size("inbound","$avp(size)"); ... } ============================================ when program runs to the function get_profile_size(), the proxy will be terminate. log file: ALTER: <core> [main.c 722] : child process 29651 exited by sinal 11 ALTER: <core> [main.c 725] : core was generated INFO : <core> [main.c 737] : terminating due to SIGCHILD ============================================ what' s the problem? Did I use the func wrong? thanks

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pcre mystery or bug in dialplan module?
by Juha Heinanen 21 Feb '11

21 Feb '11

i have in dialplan table record 287 14 10 1 ^(.+)\+358 0 ^(.+):\+358(.+) \1:0\2 and when it is applied to uri Feb 21 08:05:15 sip /usr/sbin/sip-proxy[24071]: INFO: uri <sip:+35892345670@test.fi;user=phone> the resulting uri looks like this: Feb 21 08:05:15 sip /usr/sbin/sip-proxy[24071]: INFO: uri <sip\1:092345670@test.fi;user=phone> i can't figure out, where \1 comes to the result, because \1 is has already produced 'sip' to it. i have added debug to dialplan module and looks like it compiles correct subst and repl expressions: Feb 21 08:05:05 sip /usr/sbin/sip-proxy[24067]: INFO: dialplan [dp_db.c:392]: parsing repl_exp <\1:0\2> Feb 21 08:05:05 sip /usr/sbin/sip-proxy[24067]: INFO: dialplan [dp_db.c:404]: compilig subst_exp <^(.+):\+358(.+)> is there some stupid mistake in the expressions or could it be a bug somewhere? -- juha

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decimal fraction problem
by 侯旭光 20 Feb '11

20 Feb '11

Hello I need to add q value while using function append_branch(),but the function only takes decimal fraction as the parameter. What if I want to use pv to add q value? The $var and $avp just have string and integer type. Thanks a lot!

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My new blog http://cciev.ciscovoicetech.com/
by Arun Kumar 20 Feb '11

20 Feb '11

Hi Guys, Soon, I'll be starting a new section related to Asterisk (around 4 years of full time experience with Asterisk, Trixbox, SER, OpenSer, MediaProxy, AGI*) so let me know if you like to see some topic coming. Cheers Arun

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Failure Route and Reply Routes
by Brandon Armstead 19 Feb '11

19 Feb '11

Hello, Is there any specific information / documentation as to what kind of response causes the failure_route to trigger over the reply_route. For example, it seems that 486 response triggers the failure route while as a 408 triggers the reply_route. Perhaps I am over looking something? I would imagine that 5xx and 6xx (final / non provisional responses) would trigger the failure route. While as 4xx would trigger the reply route -- however this does not always seem to be the case in the example of 486 vs 408. Any information / help in advance is greatly appreciated, thanks! Sincerely, Brandon Armstead

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Kamailio NOTIFY
by Spinov Evgeniy 19 Feb '11

19 Feb '11

Hello all. I'm having problem with sending NOTIFY packet for event dialog. The problem is, that all NOTIFY packets sent with this event is empty. ( Content-length: 0 ) and in fact, not passing actual status to the phone. In my case this is SPA962. NOTIFY packets for event presence are sent fine and status is changing normally. The same phone, works fine with Asterisk. Asterisk is sending NOTIFY with non-zero content and changing status for the subscriber as well. I've turned on pua module to send PUBLISH requests at Kamailio, it's filling DB table with subscription, everything runs fine and NOTIFY packets are coming to subscribed phone, but they are empty. I also have presence_dialoginfo module on and pua_dialoginfo module on. I've tried on Kamailio 3.0.3 and 3.1.1. Content always empty. There is no sense of pasting here packet dumps, cause everything is like should be: (UAC) SUBSCRIBE, event: presence ---> K K ---> 202 OK (SPA) SUBSCRIBE, event: dialog ---> K K ---> 202 OK PUBLISH, event: presence ---> K K ---> 200 OK NOTIFY, event: presence, Content-Length: XXX ---> UAC NOTIFY, event: dialog, Content-Length: 0 ---> SPA962 As result, softphone changes status and hardphone ( SPA962 ) - no. May be it's cause of PUBLISH event value? Please advise.

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why function result -1 cannot be assigned to variable?
by Juha Heinanen 19 Feb '11

19 Feb '11

i have a local function x that returns a positive integer or -1. when it returns -1, assignment $var(tmp) = function_x("$rd"); produces to syslog an error: Feb 19 07:32:44 sip /usr/sbin/sip-proxy[27284]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (1415,40-1415,40) i have had that function for years, but have not noticed these errors in earlier versions than 3.1. any ideas why the error is produced and how to get rid of it? -- juha

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Interfaces guestures
by Spinov Evgeniy 18 Feb '11

18 Feb '11

Hello, all. I'm having Kamailio 3.0.4. Server have 2 physical interfaces, one is private, second is public. INVITE request is coming to Kamailio from public interface and being routed by Kamailio to one of ther asterisk boxes, which is in private network. Packet is routed to correct interface, but SRC address of packet, containing public Kamailio IP. As of that, asterisk is unable to communicate properly with Kamailio. I'm not applying any packet changes in Kamailio config and cannot get why Kamailio is doing that? Kamailio is listening both networks and able to send packets with correct IP, but somewhy sending an invalid one. Am I doing something wrong? Thank you.

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Re: [SR-Users] pdb module timeouts
by Thomas Baumann 18 Feb '11

18 Feb '11

Hello Marius, thanks a lot for submitting the patch. It works like a charm now. But to be honest I had troubles to find this patch, I just checked the daily tarball [http://sip-router.org/tarballs/sr/] and found this change. How I can get patch e914f97b4 in a better way ? regards, Thomas >Hello >I have made a patch to 3.1(e914f97b4) . The poll() event list was not >cleared when a response was received so when 2 servers responded one >after the other it might block in an incorrect recv(). Can you please >test with this? >Marius ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

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Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL
by Gary Chen 18 Feb '11

18 Feb '11

Never mind. I used wrong kamailio.cfg file. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, February 17, 2011 8:51 AM To: Gary Chen Cc: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL Hello, can you send all log messages printed by: kamailio -E -ddd Thanks, Daniel On 2/17/11 2:42 PM, Gary Chen wrote: kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1 <at> m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "flags", 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Bad File Descriptor
by Daniel-Constantin Mierla 17 Feb '11

17 Feb '11

Please keep the mailing list cc-ed. My previous email had other comments/questions inline, can you answer them as well? They are relevant in troubleshooting. Thanks, Daniel On 2/17/11 7:57 PM, David J. wrote: > version: kamailio 3.1.1 (x86_64/linux) 88bda8 > flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, > USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, > SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, > FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, > USE_DST_BLACKLIST, HAVE_RESOLV_RES > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > id: 88bda8 > compiled on 04:39:27 Dec 9 2010 with gcc 4.3.2 > > > > On 2/17/11 1:46 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> what is the version you are using? >> >> On 2/17/11 7:35 PM, David J. wrote: >>> 5(20390) ERROR: <core> [udp_server.c:586]: ERROR: udp_send: >>> sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file >>> descriptor(9) >>> 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed >>> >>> >>> I see this error when I try to restart kamailio after crash; >> >> What is the cause for the crash? Do you have a log or core for that? >> >>> I see the cause of this problem is 'stale' entries in the location >>> table; >>> >>> If I delete this entries kamailio starts fine; any suggestions to >>> prevent this from continuously happening. >> Is any change in the IP address of the server upon restart? >> >> Do the errors persist or they stop after a while? They are related to >> NAT keepalives sent by nathelper module, so they don't affect the sip >> traffic and if they are for stale contacts then they are fully >> harmless. The stale contacts should be automatically removed after a >> while by usrloc module. >> >> Cheers, >> Daniel >> > -- Daniel-Constantin Mierla http://www.asipto.com

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Bad File Descriptor
by David J. 17 Feb '11

17 Feb '11

5(20390) ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file descriptor(9) 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed I see this error when I try to restart kamailio after crash; I see the cause of this problem is 'stale' entries in the location table; If I delete this entries kamailio starts fine; any suggestions to prevent this from continuously happening. Thanks.

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Kamailio 1.5.5 No TLS Segmentation Fault
by Stagg Shelton 17 Feb '11

17 Feb '11

Hello, We have been having a problem with Kamilio faulting and dumping core files on occasion. I have not been able to reproduce the failure at will, but notice the back trace seems to point toward actions with the dialogue. Below is from a backtrace of a core file from just a few minutes ago. Can anyone determine what may have caused the system to error and stop processing? Thanks Stagg Core was generated by `/sbin/kamailio -m 512'. Program terminated with signal 11, Segmentation fault. #0 0x00007f8a11d55fa7 in unref_dlg (dlg=0x7f89f7e07470, cnt=1) at dlg_hash.c:474 474 d_entry = &(d_table->entries[dlg->h_entry]); Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.5-5.fc11.x86_64 db4-4.7.25-11.fc11.x86_64 e2fsprogs-libs-1.41.9-2.fc11.x86_64 elfutils-libelf-0.147-1.fc11.x86_64 glibc-2.10.2-1.x86_64 keyutils-libs-1.2-5.fc11.x86_64 krb5-libs-1.6.3-31.fc11.x86_64 libacl-2.2.49-3.fc11.x86_64 libattr-2.4.43-3.fc11.x86_64 libcap-2.16-4.fc11.1.x86_64 libconfuse-2.6-2.fc11.x86_64 libgcc-4.4.1-2.fc11.x86_64 libselinux-2.0.80-1.fc11.x86_64 lm_sensors-3.1.0-1.fc11.x86_64 lua-5.1.4-3.fc11.x86_64 mysql-libs-5.1.46-1.fc11.x86_64 net-snmp-libs-5.4.2.1-13.fc11.x86_64 nspr-devel-4.8.4-1.3.fc11.x86_64 nss-devel-3.12.6-1.2.fc11.x86_64 nss-softokn-freebl-3.12.6-1.2.fc11.x86_64 openssl-0.9.8n-1.fc11.x86_64 pcre-7.8-2.fc11.x86_64 perl-libs-5.10.0-82.fc11.x86_64 popt-1.13-5.fc11.x86_64 radiusclient-ng-0.5.6-4.fc11.x86_64 rpm-libs-4.7.2-1.fc11.x86_64 tcp_wrappers-libs-7.6-55.fc11.x86_64 xz-libs-4.999.9-0.1.beta.20091007git.fc11.x86_64 zlib-1.2.3-22.fc11.x86_64 (gdb) bt full #0 0x00007f8a11d55fa7 in unref_dlg (dlg=0x7f89f7e07470, cnt=1) at dlg_hash.c:474 d_entry = 0x0 __FUNCTION__ = "unref_dlg" #1 0x00007f8a11d5180f in unref_dlg_from_cb (t=0x7f89f7d9c660, type=4096, param=0x7f8a1836a6e0) at dlg_handlers.c:622 dlg = 0x7f89f7e07470 #2 0x00007f8a18138ea3 in run_trans_callbacks (type=4096, trans=0x7f89f7d9c660, req=0x0, rpl=0x0, code=0) at t_hooks.c:240 cbp = 0x7f89f7dc30e8 backup = 0x71a9d0 trans_backup = 0xffffffffffffffff __FUNCTION__ = "run_trans_callbacks" #3 0x00007f8a181273cc in free_cell (dead_cell=0x7f89f7d9c660) at h_table.c:132 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x7fff4282f190 p = 0x7f89f7d3b068 #4 0x00007f8a18127bb6 in free_hash_table () at h_table.c:345 p_cell = 0x7f89f7d9c660 tmp_cell = 0x0 i = 4075 #5 0x00007f8a181342a4 in tm_shutdown () at t_funcs.c:109 __FUNCTION__ = "tm_shutdown" #6 0x00000000004529f6 in destroy_modules () at sr_module.c:321 t = 0x7349d0 foo = 0x734910 __FUNCTION__ = "destroy_modules" #7 0x000000000041f6b4 in cleanup (show_status=1) at main.c:331 No locals. #8 0x0000000000420597 in handle_sigs () at main.c:517 chld = 0 chld_status = 134 i = 12 do_exit = 1 ---Type <return> to continue, or q <return> to quit--- shutdown_time = 60 __FUNCTION__ = "handle_sigs" #9 0x00000000004217b5 in main_loop () at main.c:859 chd_rank = 12 i = 4 pid = 21442 si = 0x0 __FUNCTION__ = "main_loop" #10 0x0000000000423410 in main (argc=3, argv=0x7fff4282f498) at main.c:1321 cfg_log_stderr = 0 cfg_stream = 0x1fe1010 c = -1 r = 0 tmp_len = 0 port = 0 proto = 4910128 ret = -1 rfd = 4 tmp = 0x7fff4282ff8a "" options = 0x4b77e0 "f:cCm:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:" rand_source = 0x4b7d9c "/dev/urandom" seed = 3628387751 __FUNCTION__ = "main" (gdb) (gdb) quit

3 7

Failover with UCARP and Monit
by Hugh Waite 17 Feb '11

17 Feb '11

Hello, I am setting up a high-availablilty kamailio system using UCARP to failover between active and standby instances. To detect failure, we intend to use Monit. Monit can monitor the kamailio PID and start the process when needed (Example on the wiki http://www.kamailio.org/dokuwiki/doku.php/install:configure-initd-script) and it can also do OPTIONS pings to verify it is working. If the pings fail we will initiate a ucarp swap. However if the server is currently in standby, it does not have the V-IP address, so I don't want to run the OPTIONS pings (I think). Does anyone use a similar system and can provide an example of how ucarp, monit and kamailio can work together? Many thanks, Hugh Waite

4 5

Kamailio and Asterisk Realtime Integration using Asterisk Database
by Pavel Miskov 17 Feb '11

17 Feb '11

Hi all, I am trying to integrate Kamailio and Asterisk as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb document but I have some problems. 1. When Asterisk sends the call back to Kamailio, Asterisk generates new INVITE but with wrong From URI. In From header, "Display name" is OK (Phone A) but "From URI" is wrong (it is URI of Phone B). 2. When Phone B answers the call, Asterisk generates another INVITE to Phone B. When Phone B sends OK to this second INVITE Asterisk generates another INVITE to Phone A. 3. When Phone B sends BYE, Asterisk generates INVITE to Phone A and when it receives OK from Phone A it sends BYE to Phone A. I can fix the first problem with transformation in Kamailio but is this the way how it is supposed to work or I misconfigured something? I have tried this with kamailio 3.1 and asterisk 1.6.2.16.1 (and 1.8.2.3). Thanks Pavel

2 1

Mediaproxy 2.4.4 sessions
by Ricardo Martinez 17 Feb '11

17 Feb '11

Hello. I’m using kamailio 3.1.2 and mediaproxy 2.4.4 . Is there a way to see the active media sessions like in the old mediaproxy? I was using the command ./sessions Thanks Ricardo-.

2 1

Kamailio 3.0.4: Segfault at 4 [..] error 4 in dialog.so
by Klaus Feichtinger 17 Feb '11

17 Feb '11

Hello, does anybody have an idea, what this error messages could mean? Feb 16 11:23:27 SipSrvLstI1 kernel: [429767.707165] kamailio[10758]: segfault at 4 ip b78ba6e5 sp bfacece0 error 4 in dialog.so[b78a0000+32000] Feb 16 13:40:42 SipSrvLstI1 kernel: [438003.071797] kamailio[24880]: segfault at 4 ip b79416e5 sp bff7d460 error 4 in dialog.so[b7927000+32000] Feb 16 18:47:25 SipSrvLstI1 kernel: [456406.166885] kamailio[20939]: segfault at 4 ip b78d56e5 sp bfd821c0 error 4 in dialog.so[b78bb000+32000] I am setting the dialog flag for any incoming call..... Whenever a load test is running, this error occurs. I am not sure if it is reproduceable, or not. But I think it should be, because in worst case (during a load test over night) this error was indicated in a period of 1-2 minutes. But I've found this error only yesterday, because the server was down (as a result of this error....). The daemon (kamailio_3.0.4+lenny1_i386.deb) is running on a virtual machine (ESX) with OS Debian Lenny. How can I find / deliver additional info to this error / fault for analyzation? Thanks for any hints. Klaus

2 1

ACC og ACC_RADIUS module
by Morten Isaksen 17 Feb '11

17 Feb '11

Hi, We have a OpenSER 1.1 platform running with radius accounting and I am in the progress of updating it to Kamailio 3.1. I am trying to decide if I should do accounting via Radius or directly to MySQL on the new platform. The only benefits a can see with Radius is that you can build some redundancy into your radius client. If one Radius server is failing then try the next and you can configure radius to log to a file if the DB is down. But i think you can get the same level of redundancy with a replicated DB setup with heartbeat/pacemaker. If I choose to do the accounting direct to MySQL I will skip the Radius layer (and one error source). Are there any other pros and cons? -- Morten Isaksen

2 1

ERROR: failover functions used, but AVPs paraamters required are NULL
by Gary Chen 17 Feb '11

17 Feb '11

kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1 <at> m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "flags", 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary

2 1

Re: [SR-Users] pdb module timeouts
by Thomas Baumann 16 Feb '11

16 Feb '11

2 1

Re: [SR-Users] Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 16 Feb '11

16 Feb '11

Hi Marius, Thanks for the reply. I found the patch in the 1.5 CVS branch. By the way, the 1.5.5 source tarball in /pub/kamailio/1.5.5/src was created before this patch was written, so the patch is not in there. The tarball I created from the 1.5 branch did have it. Thanks again, Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Mon, 14 Feb 2011, marius zbihlei (marius.zbihlei(a)1and1.ro) wrote: On 02/11/2011 09:14 PM, Sean O'Donnell wrote: > Hi all, > Hello Sean, > I checked the GIT repository and noticed that there was a patch in forward.c for > this issue. Looks like it was done 11/4/2010. Two questions: > > 1) Is there any reason that patch didn't make it into Kamailio 3.1.2 ? > Indeed, there is no reason. Actually I have cherry-picked the patches into the 3.1 branch (commit_notice 3.1 3ec1e9 and 9ef0a1e). Thanks for noticing. > 2) Any reason I shouldn't apply that patck to my 1.5.4 release? > The patch was already applied to the svn 1.5 branch (don't know if the latest 1.5.5 tar ball has it) but you can install it from svn and you can find it there (commit 6050) Cheers Marius > Thanks, > Sean O'Donnell > Senior Engineer > uReach Technologies, Inc. > > > > > > ---- On Thu, 10 Feb 2011, Sean O'Donnell (skodonnell(a)ureach.com) wrote: > > Hi all, > > I just started deploying Kamailio release 1.5.4, and I think there's an issue > with how Kamailio identifies an outgoing interface when mhomed is enabled under > Linux. > > I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a > voicemail platform. It > runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is > configured such that it listens on both interfaces. One interface (public > interface) handles traffic with the SBC, the other (private interface) handles > with the VM platform. The 'mhomed' option is enabled. > > After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP > packets coming out of the public interface. After looking at some ngrep > captures on that interface, I noticed that some packets had the source IP > address of the private interface and also had Record-Route and Via headers for > the private interface only - no headers for the public interface were there. > > Usually when I see the wrong source IP in a UDP packet, it's an issue with how > routes are set up on the host. However, I had our network engineer double check > them, and they seem fine (no ambiguous routes). The fact that I captured these > messages on the public interface also indicates to me that the kernel is routing > the message correctly. The missing Record-Route and Via for the public > interface, however, lead me to believe that the proxy didn't correctly identify > the outgoing interface in the first place. > > After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was > put in to improve performance when mhomed is enabled (r5971) in forward.c, and I > think this is the issue. > > As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined > the outgoing interface by creating a temporary UDP socket, invoking connect() on > the socket with the packet destination, then checking the source IP of the > socket that the kernel assigned using getsockname(). After the source address > was determined, the temp socket was closed closed. As of 1.5.4, this was > modified to reuse the temporary socket and just re-invoking connect() with a new > destination address. > > The problem with the enhancement is that Linux (again, at least in the 2.6 > kernels I'm using) > doesn't seem to rebind a new source address to the socket when connect() is > called more than once on > a UDP socket. Instead, it keeps the original one, and thus the wrong interface > is assumed. > > I wrote a small program to confirm this - basically creates a UDP socket, calls > connect()/getsockname() > multiple times using different destination addresses. I ran it on several 2.6 > kernels, including > Centos4.x and Centos5. The result was always that the source address of the > socket wasn't changed after the first connect(), regardless of the destination > address. The only way I could get it work as > required was to first do a connect() using a zero'd out AF_UNSPEC address before > doing the > connect() to the remote address. I also ran it on Solaris and it worked. Go > figure. > > I've downloaded the latest stable release (3.1.2) but I think the issue is still > there, and I don't see > anything in the user groups that addresses this. > > Any help would be appreciated. > > Thanks, > > Sean > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Error when tring to load lcr module.
by Gary Chen 16 Feb '11

16 Feb '11

I am using Kamailio 3.1.2. I added lcr module in my kamailio.cfg file. After run this: ./kamctl -E I got this error: 0(23744) ERROR: lcr [lcr_mod.c:1235]: lcr_gw params at row <0> does not start with ';' 0 I have set up some gateway information in lcr_gw , lcr_rule and lcr_rule_target. From log it seems retrieve all the information but then got above error. Here is my lcr part of config: loadmodule "lcr.so" modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") Gary

2 1

Re: [SR-Users] pdb module timeouts
by Thomas Baumann 15 Feb '11

15 Feb '11

Hello, I did a fallback to the original source, without sucess. I still have massive timeouts. So far I have no idea, how to go on. ps. what lenght of the carrier codes would be ok ? 3...10 digits, additional characters needed like A,B,C,D,E,F? regards, Thomas Hi Henning, this change was done at pdbt.c: $ diff pdbt.c pdbt_changed.c 245c245 < bufsize = slen + 1 + 1 + 3 + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) --- > bufsize = slen + 1 + 1 + 4 + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) 257,258c257,258 < ret = snprintf(p, 5, "%d\n", node->carrier); < if (ret < 1 || ret > 4) { --- > ret = snprintf(p, 6, "%d\n", node->carrier); > if (ret < 1 || ret > 5) { If I run querys in filemode against the server, where is no timeout. pdbt query -f /tmp/numbers3 -r 10.12.18.21:10001 -q 1 | grep answer regards, Thomas -----Ursprüngliche Nachricht----- Von: "Henning Westerholt" <henning.westerholt(a)1und1.de> Gesendet: 11.02.2011 09:33:05 An: sr-users(a)lists.sip-router.org Betreff: Re: [SR-Users] pdb module timeouts On Wednesday 09 February 2011, Thomas Baumann wrote: > I am using the PDB module and server components for number portability. 2 > instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on > (10.12.19.21). With a small amount of traffic (-cmax 150 -cps 10 > -callduration 3), where are timeouts: WARNING: pdb [pdb.c:260]: exceeded > timeout while waiting for response. > > One requested number was 307111094, where the module prints out a timeout. > > The funny part is, that I can see the responses at least arriving at the > 10.12.19.21 interface on time. > > Request send: 0,200855 s > Answer received: 0,201027 s > > That are 0,172 ms and far away from a timeout. > > What could be the reason ? Hi Thomas, a bit difficult to say on a first sight.. Maybe some scheduling or load issues on the local machine? I've just checked two production server here, i don't see it so far in the available logs. How much load do you've on the machine? And what kind of timeout value do you specified? > ps. A small change on the server part was done: handle 4 digit carrier > codes. Ah, i see. I guess its not a big change? Regards, Henning ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

2 1

CORE postgres module
by Bruno Bresciani 15 Feb '11

15 Feb '11

Hi, During my tests with kamailio 1.5.0 a core is generated when postgres is disconnected and I try register a user or make a call in kamailio. Analising the module postgres source code I notice that core is generated by the PQescapeStringConn located in db_postgres_val2str function. Someone know why this core is generated? Best Regards

2 2

Issue with avp
by Amit Nepal 14 Feb '11

14 Feb '11

Hi, I have been trying to load rpid while loading credentials. modparam("auth_db", "load_credentials", "$avp(i:123)=rpid") Now, I am trying to do a check in my routing logic. xlog("L_NOTICE","The avp is :$avp(i:123)"); (I dont get the value here either, i can't see the value when i do avp_print() if($avp(i:123)<5) { sl_send_reply("408","Message Here"); } I dont get the value of avp at that place. Any guidance please. -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 4

kamailio 1.5.5 crashes continue
by Andrew O. Zhukov 14 Feb '11

14 Feb '11

Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 6, Aborted. #0 0x00000039d8c30265 in raise () from /lib64/libc.so.6 (gdb) backtrace #0 0x00000039d8c30265 in raise () from /lib64/libc.so.6 #1 0x00000039d8c31d10 in abort () from /lib64/libc.so.6 #2 0x000000000046c397 in qm_debug_frag (qm=0x733c00, f=0x841420) at mem/q_malloc.c:137 #3 0x000000000046d99a in qm_free (qm=0x733c00, p=0x841450, file=0x4e4d30 "parser/digest/digest.c", func=0x4e4da0 "free_credentials", line=95) at mem/q_malloc.c:439 #4 0x0000000000495fac in free_credentials (_b=0x2b875404a130) at parser/digest/digest.c:95 #5 0x0000000000471a36 in clean_hdr_field (hf=0x2b875404a100) at parser/hf.c:116 #6 0x00002b8750ab78de in clean_msg_clone (msg=0x2b8754049130, min=0x2b8754049130, max=0x2b875404a748) at sip_msg.h:54 #7 0x00002b8750ab77b7 in run_trans_callbacks (type=2, trans=0x2b8754057ec0, req=0x2b8754049130, rpl=0x7c0eb8, code=200) at t_hooks.c:245 #8 0x00002b8750abe39d in t_reply_matching (p_msg=0x7c0eb8, p_branch=0x7fffbe5d11e8) at t_lookup.c:888 #9 0x00002b8750abe997 in t_check (p_msg=0x7c0eb8, param_branch=0x7fffbe5d11e8) at t_lookup.c:964 #10 0x00002b8750acd79b in reply_received (p_msg=0x7c0eb8) at t_reply.c:1395 #11 0x000000000041c6db in forward_reply (msg=0x7c0eb8) at forward.c:576 #12 0x000000000043ccf0 in receive_msg ( buf=0x712980 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP XXX.XX.XXX.13;branch=z9hG4bK8aa9.93e271f1.0;received=XXX.XX.XXX.13\r\nVia: SIP/2.0/UDP XXX.XX.XXX.236:5060;received=XXX.XX.XXX.236;branch=z9hG4bK37dacff3;rport=5060\r\nRec"..., len=576, rcv_info=0x7fffbe5d1340) at receive.c:212 #13 0x00000000004692e3 in udp_rcv_loop () at udp_server.c:449 #14 0x0000000000420ecb in main_loop () at main.c:774 #15 0x0000000000422e0f in main (argc=11, argv=0x7fffbe5d15c8) at main.c:1321 (gdb) q [root@ tmp]# openser -V version: kamailio 1.5.5-notls (x86_64/linux) flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 09:42:37 Feb 10 2011 with gcc 4.1.2 Andrew.

2 3

kamailio and MySQL licensing question
by Siddhardha Garige 14 Feb '11

14 Feb '11

hello all, I have a question regarding MySQL licensing. We configured kamailio to work in edge proxy mode with drive-by registration to support our network architecture. One of our clients is interested in conducting some trails with us and we are setting up a test network at client's facility. As a part of this process we will be installing our custom kamailio on their servers with MySQL. We are not charging our client for kamailio and mysql implementation since they both are open source implementations. But we will be charging them for our time at clients facility. Are we violating any MySQL license policies by installing our custom kamailio with MySQL for our client? thanks Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

2 1

Re: [SR-Users] Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 14 Feb '11

14 Feb '11

Hi all, I checked the GIT repository and noticed that there was a patch in forward.c for this issue. Looks like it was done 11/4/2010. Two questions: 1) Is there any reason that patch didn't make it into Kamailio 3.1.2 ? 2) Any reason I shouldn't apply that patck to my 1.5.4 release? Thanks, Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Thu, 10 Feb 2011, Sean O'Donnell (skodonnell(a)ureach.com) wrote: Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Error running kamailio 3.1.0
by Lucas Alvarez 12 Feb '11

12 Feb '11

Hi, I have compiled kamailio 3.1.0 without any error and I having this error when running kamailio: ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/sl.so>: /usr/local/lib/kamailio/modules_k/sl.so: undefined symbol: fm_malloc Any will be appreciated. Regards, Lucas

2 1

After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set
by Andrew O. Zhukov 12 Feb '11

12 Feb '11

[root@ tmp]# /usr/local/sbin/kamailio -V version: kamailio 1.5.5-notls (x86_64/linux) flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 12:38:36 Feb 2 2011 with gcc 4.1.2 ----------------------------- Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 11, Segmentation fault. #0 0x000000000046b0e3 in fm_malloc (qm=0x72dc00, size=32) at mem/f_malloc.c:354 354 if ((*f)->size>=size) goto found; (gdb) backtrace #0 0x000000000046b0e3 in fm_malloc (qm=0x72dc00, size=32) at mem/f_malloc.c:354 #1 0x00002b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80, user=0x7fffe9c5a500, tag=0x777a58, params=0x0, _inbound=0) at record.c:176 #2 0x00002b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at record.c:322 #3 0x00002b30f28047db in w_record_route (msg=0x76e0e0, key=0x0, bar=0x0) at rr_mod.c:212 #4 0x000000000040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at action.c:874 #5 0x000000000040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) at action.c:145 #6 0x000000000040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at action.c:746 #7 0x000000000040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) at action.c:145 #8 0x000000000040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at action.c:120 #9 0x000000000040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at action.c:195 #10 0x000000000043bda4 in receive_msg ( buf=0x70c980 "NOTIFY sip:XXXXXX.com SIP/2.0\r\nVia: SIP/2.0/UDP XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome <sip:101650@XXXXXX.com>;tag=129d73a13db8ec7fo0\r\nTo: <sip:XXXXX.com>\r\nCall-ID: e3fd1da9-142a0a17"..., len=373, rcv_info=0x7fffe9c5ae90) at receive.c:175 #11 0x0000000000467eeb in udp_rcv_loop () at udp_server.c:449 #12 0x000000000042097b in main_loop () at main.c:774 #13 0x00000000004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321 (gdb) print size $1 = 32 (gdb) quit -------------------------------------------- Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 11, Segmentation fault. #0 0x000000000046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609 609 size+=f->size,f=f->u.nxt_free,i++,j++){ (gdb) backtrace #0 0x000000000046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609 #1 0x000000000041feb3 in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x0000000000467bf4 in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042097b in main_loop () at main.c:774 #6 0x00000000004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321 (gdb) print i $1 = 402 (gdb) print j $2 = 1 (gdb) print size $3 = 7234295468789601279 (gdb) print f $4 = (struct fm_frag *) 0x3738656435393838 (gdb) print f->size Cannot access memory at address 0x3738656435393838 ------------------------------------------------------------------- Andrew O. Zhukov

2 6

pdb module timeouts
by Thomas Baumann 11 Feb '11

11 Feb '11

Hello, I am using the PDB module and server components for number portability. 2 instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on (10.12.19.21). With a small amount of traffic (-cmax 150 -cps 10 -callduration 3), where are timeouts: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response. One requested number was 307111094, where the module prints out a timeout. The funny part is, that I can see the responses at least arriving at the 10.12.19.21 interface on time. Request send: 0,200855 s Answer received: 0,201027 s That are 0,172 ms and far away from a timeout. What could be the reason ? regards, Thomas ps. A small change on the server part was done: handle 4 digit carrier codes. level3_logs: Feb 9 17:19:42 node2 /openser/sbin/kamailio[21200]: DEBUG: pdb [pdb.c:201]: querying '307111094'... Feb 9 17:19:42 node2 /openser/sbin/kamailio[21200]: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response. ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

3 4

Change transport=tls within Contact Header
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, is there a function to change the transport field within the Contact Header? t_relay does relay it to the UDP port of the PBX but the transport within the Contact Header is still set to TLS. Later, the PBX does try to establish a TLS connection. The idea is now, that kamailio does set the transport of the Contact Header to UDP. Another possible way would to be, that I use the re.subst method but maybe there is a special function for that. Thanks in advance. Best regards, Bernhard

2 1

NAT, TLS and location table
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, I am using TLS and recognize the following problem: The TLS connection are build up successfully but the natping (natping_interval = 10) does not send small dummy packets to the phones. The phones are behind a firewall with NAT. Registered phones with NAT but UDP do work correctly. They are getting the natping every 10 seconds. After 120 seconds (should be the tcp_connection_timeout) kamailio does send a FIN to the TLS phone to close the TLS connection. Should I increase the tcp_connection_timeout to a value bigger than the registration timeout? I thought I do not need that, because of the natping_interval. Is it maybe better to use a SIP-Options Ping instead of the small dummy packets? I would prefer the dummy packets because they are much smaller. BTW: Kamailio 3.1.1 and snom360 with latest v8 firmware. Thanks in advance! Best regards, Bernhard

2 3

handle_new_connect ... Too many open files
by Efelin Novak 11 Feb '11

11 Feb '11

Hi, today during two hours kamailio made 2G of a syslog messages. This message was repeated for 10 000 000 times. /sbin/kamailio[24272]: ERROR: <core> [tcp_main.c:3959]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files It was preceded by following error. /sbin/kamailio[24265]: ERROR: <core> [tcp_read.c:269]: error reading: Connection reset by peer (104) /sbin/kamailio[24265]: ERROR: <core> [tcp_read.c:882]: ERROR: tcp_read_req: error reading Finally it stopped without any given reason. I'm using kamailio 3.1.0 (i386/linux) 1e204f, callcontrol 2.0.11 and clustered mysql. Was this some kind of attack, bug, error, misconfiguration or what? Thank for answer, Efelin

2 1

TLS Support
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, how could I compile the TLS module and only the TLS module - not the other sources. BTW: The documentation within the http://www.kamailio.org/docs/tls.html is not up-to-date. Thanks in advance. Best regards, Bernhard

2 2

Kamailio as a sip client
by Amit Nepal 11 Feb '11

11 Feb '11

Hi, Is it possible to have kamailio register to a different proxy and route calls through that proxy ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 3

Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 10 Feb '11

10 Feb '11

Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean

1 0

Merging nathelper modules
by Ovidiu Sas 10 Feb '11

10 Feb '11

Hello all, The nathelper module in modules_k was split in two: - nathelper (dealing with signaling); - rtpproxy (dealing with rtpproxy protocol). I would like to move the rtpproxy module from "modules_k" into "modules" and remove rtpproxy functionality from nathelper (s). This will give to (s) and (k) users: - rtpproxy: a single module for dealing with rtpproxy servers; - nathelper: two variants for dealing with NAT signaling. Next step, will be to merge the two nathelper modules into a single one. Thoughts? Regards, Ovidiu Sas

2 2

nathelper-module and sdp offer in 200 OK
by Emil Kroymann 10 Feb '11

10 Feb '11

Hi, We recently had a problem with the nathelper module and rtpproxy in a scenario where the SDP offer is sent only in the 200 OK. We use sip-router 3.1 and rtp-proxy from git master. The sip-router configuration uses the rtpproxy_offer() and rtpproxy_answer() functions in appropriate places. The problem is, that the arguments sent to the rtpproxy, when the ACK with the sdp answer arrives, seems to be not in the order, that rtpproxy expects. On the 200 OK, the nathelper module sends callid, to-tag, from-tag to rtpproxy. On the ACK, the nathelper module sends callid, from-tag, to-tag (with different command prefixes, of course, but I cannot remember them atm). The version of rtpproxy that we are using seems to expect, that the order of arguments sent on the ACK request is the same as on the 200 OK. My question: are there any module parameters, to correct this behaviour? Regards, Emil -- Emil Kroymann VoIP Services Engineer Email: emil.kroymann(a)isaco.de Tel: +49-30-203899885 ISACO GmbH Kurfürstenstraße 79 10787 Berlin Germany Amtsgericht Charlottenburg, HRB 112464B Geschäftsführer: Daniel Frommherz

3 12

Use different certificate for different client with TLS
by Daniel GROTTI 09 Feb '11

09 Feb '11

Hi all, I would like to use kamailio 3.1 with TLS and verified also a client certificate. My tls.cfg file is as follow: --- tls.cfg ---- ..... ..... [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key.pem certificate = default_cert.pem ca_list = default_ca.pem [client:default] verify_certificate = no require_certificate = no .... .... So I want to verify the client certificate, to do that I use "is_peer_verified()" function in kamailio.cfg. As tls.cfg shows, I have to send to my clients the CA certificate and the client certificate (default_cert.pem + default_key.pem - signing by the CAcert). This client certificate is unique for all clients. Everything works fine. But suppose I wanted to create a client certificate for client 1 (cert_1.crt), and a different client certificate for client 2 (cert_2.crt) and I want to configure kamailio to be able to verified this different certificates. Does it possible ? How can I configure the tls.cfg file to do that ? I try to do something like this: [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key_1.pem certificate = default_cert_1.pem private_key = default_key_2.pem certificate = default_cert_2.pem ca_list = default_ca.pem But when kamailio restart it seems that it read only the last couple of row certifcate/private_key. Regards, Daniel G

2 1

[OpenSER-Users] uaCSTA support in openser
by Hugo Koblmueller 09 Feb '11

09 Feb '11

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

7 13

Re: [SR-Users] module postgres SER
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

please keep the mailing list cc-ed, sending private messages is not in the spirit of public mailing lists. Others may want to follow up the discussion now or later. Thanks, Daniel On 2/9/11 5:01 PM, Bruno Bresciani wrote: > Daniel, > > thanks for your reply, really ser-0.8.1.4 is too old but i need to > solve this problem on that version. My great doubt is Why the aug_free > function corrupt the url of database after some attemps to reconnect. > Well, I'll try to understand this question... > > Best Regards > > > 2011/2/9 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 2/9/11 2:19 PM, Bruno Bresciani wrote: > > Hi, > > I've seen the problem in a postgres module (SER-0.8.1.4), if > the connection fails and module tries to reparse url it fails > as CON_SQLURL(_h) is corrupted by the function aug_free() > after some reconnect attempts . When the postgres database > back to work, some modules doesn't get reconnect because the > db_url is corrupted. Why this is happening? There are some > solution for this problem? > > ser 0.8.1.4 is soooo old and I cannot fully remember, but I think > postgres module had no reconnect functionality at all by that time. > > However, version 3.1.x of SER (as well as Kamailio flavour) has db > reconnect functionality for postgres. You can try it and report if > something is not working, it will be fixed in 3.x, but I think > nobody is still developing on 0.8.x to backport anything there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

problem with bye using rtpproxy
by Amit Nepal 09 Feb '11

09 Feb '11

I have been trying to figure this out While using kamailio and rtpproxy, the caller is not receiving the bye when callee hangs up but audio is two way and everything seems to be working fine, any one had this issue ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 1

Multiple call accounting from an IP
by Amit Nepal 09 Feb '11

09 Feb '11

Hi everyone, I am sure someone have been working with this scenario, how about accounting multiple calls from same account or ip address while using ip auth ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 1

fosdem 2011 presentation about p_usrloc
by Henning Westerholt 09 Feb '11

09 Feb '11

Hello, this year the FOSDEM developer conference was a again a really nice event, the first time with an own room dedicated completely to open source telephony solutions! If you're interested in our presentation about the new p_usrloc module and how to scale location services with Kamailio, you can find it at the usual place on our webserver: http://kamailio.org/events/2011-fosdem/p_usrloc.pdf Best regards, Henning

3 3

module postgres SER
by Bruno Bresciani 09 Feb '11

09 Feb '11

Hi, I've seen the problem in a postgres module (SER-0.8.1.4), if the connection fails and module tries to reparse url it fails as CON_SQLURL(_h) is corrupted by the function aug_free() after some reconnect attempts . When the postgres database back to work, some modules doesn't get reconnect because the db_url is corrupted. Why this is happening? There are some solution for this problem? Best Regard

2 1

Fosdem2011 presentation: SIP Web20 Lua
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

Hello, the presentation I did during Fosdem in Brussels last weekend is available at: http://www.kamailio.org/events/2011-fosdem/dcm-sip-web-lua.pdf The focus was to show how to interact with Kamailio via HTTP and how Kamailio can interact with Web services via HTTP, using Lua to make it easier. There are slides for a demo config of sending asynchronous notifications to Twitter on missed calls (using modules app_lua, mqueue, sqlops and rtimer). Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] problem with SER not proxying messages
by Asuncion Merino Rodríguez 09 Feb '11

09 Feb '11

Hola, no he visto respuesta a este foro, por favor alguien podría dar alguna respuesta?

1 0

Kamailio v3.1.2 Released
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, Kamailio SIP Server v3.1.2 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.1. There is no change to database or configuration file required to upgrade to 3.1.2 from 3.1.0 or 3.1.1 versions, therefore it is strongly recommended to upgrade to v3.1.2. For more details about version 3.1.2, visit: http://www.kamailio.org/w/2011/02/kamailio-v3-1-2-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Problem with LCR!
by Ricardo Martinez 08 Feb '11

08 Feb '11

Hello. We’re having critical issues with our LCR, we have the Kamailio version 1.5.2, several calls are arriving to our LCR and from some reason the next message appears in the log : Feb 6 12:01:55 /usr/local/sbin/kamailio[14185]: NOTICE:lcr:gw_set_state: trunk "xx.xx.xx.135:5060" from group: <2099> is OFFLINE! We can see this message repeated for every single gateway in the “gw” table from the LCR, after this happened, the LCR rounting process is blocked and no more calls are processed. Can someone help us with this problem? Thanks in advance, *Regards,* *Ricardo Martinez.-* * *

3 3

planning release of v3.1.2
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, I think it is time to release v3.1.2, first date that comes in my mind is next Thursday if everyone feels it is enough time to take care of backporting any fix he/she did and it is not yet there. That will provide us a fresh release for the FOSDEM event. If not, then maybe the other week, Tuesday, so the participants at the Kamailio Devel training in Barcelona can practice on it. Soon after we should plan also a release for previous stable, branch 3.0. Anyone having other options? Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

Problem with load_gws()
by Amit Nepal 08 Feb '11

08 Feb '11

Hi, Below is my config and i am getting error , missing load module modparam("lcr","db_url",DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") route[LCR] { if(!load_gws()){ xlog("yes"); }; This gives me error : loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio 0(1265) : <core> [cfg.y:3412]: parse error in config file routes/route-lcr.cfg unknown command, missing loadmodule? for load_gws() ERROR: bad config file (1 errors) Any guidance please ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 4

Handling timers from an Avaya PBX (throwing 404s)
by Graham Wooden 08 Feb '11

08 Feb '11

Hi there, I turned up a few new sip trunk customers this past week, from which one of them was an Avaya system. The customer wasn¹t able to do a username/password auth, so I set them up as a permanent entry in the location table. I am running Kamailio 1.5.5. The issue is that I am throwing back a 404 to them when they do a timer event, as they are passing sip:Unknown@my.ip.addr. I don¹t have a user name Unknown, hence the 404. I noticed that with the timer events from other customer¹s PBXs, that they are passing sip:my.addr, from which I am giving back 200 OK. What is the best way to handle this? Just let it go or add a ³Unknown² user (I don¹t like the latter, really)? Calls to and from their system is fine, so right now it¹s just entries in my logs but I hate spending CPU cycles on stuff that isn¹t going to benefit anyone. I sent an email back to the customer to see if maybe he can resolve it on his end, but I am doubtful on getting it resolved on his end. Thanks, -graham

3 5

Proposal to extend acc/dialog modules in order to log CDRs
by Timo Reimann 07 Feb '11

07 Feb '11

Hi all, I'd like to bring forward a proposal for creating CDRs (Call Data Records) from Kamailio by means of extending both acc and dialog modules. Feedback is very welcome. So far, CDR generation requires post-processing of log entries created by the (transaction-based) acc module or some other means. Looking at OpenSIPS (which is already capable of producing CDRs), my idea is to copy the basic notion of tracking dialogs for to-be-accounted calls and log a proper CDR on termination. Creating basic CDRs including start time, end time, and call duration can be accomplished quite easily this way. Apart from this very minimum CDR content, however, one can think of a number of CDR fields that cannot be filled easily. For instance, caller identity isn't always contained in the From header (think of CLIR calls) but need to be determined from other headers depending on the type of call, possibly even involving a database lookup. That's why I'd like to provide a way to pass additional, CDR-specific data from the configuration file to the dialog structure associated with the call. When the dialog is about to terminate, the extended acc module would make sure that the added CDR data associated to the dialog is included in the CDR. This way, users may define CDR fields to their personal needs from the configuration file in a flexible manner. In order to accomplish this, the dialog module would need to be extended such that dialog-specific data may be stored for the duration of a dialog (which is not possible to this day AFAICS). A new setter function attached to the dialog interface (struct dlg_binds) could accept a simple key/value pair for storage purposes. Exporting this function to the configuration file and accompanying it with a corresponding getter function will ensure that dialog-specific data may be written and read generically. Thus, improvements to the dialog module may be leveraged not just for CDR-accounting purposes. Regarding the acc module, a couple of new features need to be implemented: First, the introduction of another module parameter called something like "cdr_fields" that comprises the set of key names designated for CDR inclusion. Hence, a line like modparam("acc", "cdr_fields", "caller, callee, foo, bar") would require Kamailio configurators to store dialog-specific values for the keys "caller", "callee", "foo", and "bar", respectively, during call processing. The values will be fed to the dialog structure employing the new dialog store function explained above. Second, when the acc module is in CDR mode, dialogs shall be tracked from creation to termination using dialog callbacks. On termination of a dialog, values from keys referenced in the "cdr_fields" parameter will be fetched from the terminating dialog structure and associated with the new, fresh CDR. Third, the CDR is persisted to either log file, database, or both. Note that my approach would not touch the existing acc module functionality: CDR mode is supposed to be working in addition to transaction-based accounting as people will likely want to keep transaction logs for debugging or similar needs. A flag would enable and disable CDR generation at will. I have not fully investigated re-usability possibilities but believe that acc's persistence functions may be taken advantage of by the new CDR-producing parts as well. That's it with regards to the basic mode of operation. I am also thinking of having a few other features, such as using a switch to set the call start time to the moment the ACK or OK is seen instead of the INVITE (because users shouldn't be billed for call setup times). Looking forward to seeing your feedback on this proposal. Cheers, --Timo

3 6

topoh, angle brackets and Contact URI params interpretation
by Andrew Pogrebennyk 07 Feb '11

07 Feb '11

I'm having an issue with topoh module in Kamailio 3.1.0. When Contact header is formed like this: Contact: 0991 <sip:192.168.0.107;line=sr-N6IAzBysz.tyz.D4M.VLOBMfOBFuWxvfMxV4> The other party responds properly. But when there is no angle brackets in Contact: Contact: sip:192.168.0.107;line=sr-N6IAzB3AWxyfz.stM.quOBFZMJZfWxj7W.y-MljAWBy* Really, many parsers implement "prefer shift to reduce" principle, which means if something can be interpreted in more enclosed expression, it will be interpreted this way and no as part of less enclosed expression so ;line is interpreted as header parameter but not URI parameter. RFC 3261 section 20 suggests that any URI parameters be contained within angle brackets: === The Contact, From, and To header fields contain a URI. If the URI contains a comma, question mark or semicolon, the URI MUST be enclosed in angle brackets (< and >). Any URI parameters are contained within these brackets. If the URI is not enclosed in angle brackets, any semicolon-delimited parameters are header-parameters, not URI parameters. === I think the topoh module should force the angle brackets. BTW it seems that parameter needs to be urlencoded, see rule 'other-param' in RFC 3261 section 25.1: other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$" No .-* characters are allowed in the paramchar. But at least that's not causing me any problems. -- Sincerely, Andrew Pogrebennyk

3 6

Compiling Just One Module
by Amit Nepal 07 Feb '11

07 Feb '11

Could someone guide me how to compile and install just one module with flavor set to kamailio ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 4

authentication is not working
by Danny Dias 06 Feb '11

06 Feb '11

Hello my friends, I'm trying to configure authentication on my Kamailio and is not working at all :( I've added the following to the script to make it work: (but it doesn't) ... loadmodule "auth.so" loadmodule "auth_db.so" ... modparam("usrloc", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("auth_db", "load_credentials", "") ... if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/ { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!db_check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); } ... if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } ## if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } But is not working at all...take a look: # U 2011/02/03 09:31:04.402891 172.30.140.22:48752 -> 172.30.140.8:5060 REGISTER sip:172.30.140.8 SIP/2.0 Via: SIP/2.0/UDP 172.30.140.22:48752 ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport Max-Forwards: 70 Contact: <sip:1000@172.30.140.22:48752;rinstance=fcade2df86ce0ab8> To: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>> From: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>>;tag=cd3e2323 Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 # U 2011/02/03 09:31:04.404039 172.30.140.8:5060 -> 172.30.140.22:48752 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.140.22:48752 ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport=48752 To: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8> >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.34dc From: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>>;tag=cd3e2323 Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc. CSeq: 1 REGISTER Contact: <sip:1000@172.30.140.22:48752 ;rinstance=fcade2df86ce0ab8>;expires=3600 Content-Length: 0 Am i missing something in my configuration? Thanks in advance!!!

5 12

503 Server error response for REGISTER request (Asterisk v1.6.0.5)
by Vikram Ragukumar 04 Feb '11

04 Feb '11

Hello, I had posted this question earlier on asterisk-users, but didn't receive any reply, so I'm posting on sr-users in the hope that someone can provide guidance on how to debug this problem. Our setup involves a Sip softphone registering with a fresh install of Asterisk v1.6.0.5 through an outbound proxy (Kamailio v3.1.1). We observe that about 80% of the time, immediately after startup of the softphone, when the softphone sends a REGISTER message, Asterisk replies with a "503 Server error". However, subsequent REGISTER messages are replied to appropriately (with a 401 Unauthorized). There are very few devices (under 5) registered with Asterisk, so this is not an overload issue. Please find attached a SIP trace for device registration. - What might cause this behavior in Asterisk ? - Are there any config file parameters that require to be changed ? - Would it help to see any other specific logs, if so which ones ? Thanks and Regards, Vikram.

2 3

SIP Router 3.03 topoh
by dotnetdub 04 Feb '11

04 Feb '11

Hi All, Kamailio had just started to get SIP attacked and it looked like they were sending a bad header and Kamailio died.. Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [R-MAIN =>] Request method OPTIONS is to our domain (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1] Received registration from sip:1368250498@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1] Received registration from sip:100@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3374]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.401446] kamailio[3374]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3373]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.636774] kamailio[3373]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3388]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.680305] kamailio[3388]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3390]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.710444] kamailio[3390]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1] Received registration from sip:2478223614@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3378]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.753334] kamailio[3378]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1] Received registration from sip:noauth@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3375]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.768137] kamailio[3375]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.775628] kamailio[3383]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3386]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.790925] kamailio[3386]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.798926] kamailio[3379]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3381]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.815493] kamailio[3381]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:54 sipproxy sip[3344]: INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD Nov 20 19:48:54 sipproxy sip[3399]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3396]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3400]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3392]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3401]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3394]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3402]: INFO: <core> [main.c:788]: INFO: signal 15 received Thanks Brian

4 20

dispatcher - ds_mark_dst("i"); is inactive only few seconds
by Thomas Baumann 04 Feb '11

04 Feb '11

Dear sr-users, I use this straight forward dispatcher config. In the failure route I am calling this function: ds_mark_dst("i"); Also this action is logged: Feb 2 17:20:31 node5 /openser/sbin/kamailio[15945]: INFO: : FAIL_ONE:time [Wed Feb 2 17:20:31 2011] destination sip:10.12.19.31:5060 marked as inactive The gateway is inactive: [root@node5 ~]# kamctl 'fifo' ds_list URI:: sip:10.12.19.31:5060 flag=I priority=0 attrs=test1=123;test2=456 URI:: sip:10.12.19.21:5060 flag=A priority=0 attrs=test1=123;test2=456 Few seconds later the gateway is active again (there is still now response to SIP-OPTIONS) [root@node5 ~]# kamctl 'fifo' ds_list URI:: sip:10.12.19.31:5060 flag=A priority=0 attrs=test1=123;test2=456 URI:: sip:10.12.19.21:5060 flag=A priority=0 attrs=test1=123;test2=456 Does anybody have a idea why ? regards, Thomas DB: mysql> select * from dispatcher -> ; +----+-------+----------------------+-------+----------+---------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+----------------------+-------+----------+---------------------+-------------+ | 1 | 1 | sip:10.12.19.21:5060 | 0 | 0 | test1=123;test2=456 | AS 1 | | 2 | 1 | sip:10.12.19.31:5060 | 0 | 0 | test1=123;test2=456 | AS 2 | +----+-------+----------------------+-------+----------+---------------------+-------------+ CONFIG: #!KAMAILIO ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_IPAUTH # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@10.12.18.51/openser" #!endif debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="m-lab-ca805-sig.kd-lab.de:5060" alias="m-lab-ca805-sig.kd-lab.de" alias="10.12.19.51" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.12.19.51:5060 listen=tcp:10.12.19.51:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 ####### Modules Section ######## # set paths to location of modules mpath="/usr/local/kamailio-3.1/lib/kamailio/modules_k/:/usr/local/kamailio-3.1/lib/kamailio/modules/" # ------------------ module loading ---------------------------------- 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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dialplan.so" loadmodule "avpops.so" loadmodule "dispatcher.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- dispatcher params ----- #modparam("dispatcher", "list_file", "/openser/etc/kamailio/dispatcher.list") modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1@m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; #!ifdef WITH_IPAUTH if (!allow_source_address()) { sl_send_reply("403", "Forbidden"); exit; } #!endif # Send to application servers for number evaluation if ($rU=~"^0[1-9][0-9]{5,20}$") { ds_select_dst("1","0"); if (method=="INVITE") { t_on_failure("FAIL_ONE"); } t_relay(); exit(); } sl_send_reply("403", "Forbidden"); exit; } # Sample failure route failure_route[FAIL_ONE] { if (t_check_status("408")) { xlog("L_INFO", "FAIL_ONE:time [$Tf] destination $avp(dsdst) marked as inactive \n"); ds_mark_dst("i"); } exit; } ___________________________________________________________ NEU: FreePhone - kostenlos mobil telefonieren und surfen! Jetzt informieren: http://produkte.web.de/go/webdefreephone

2 5

Reloading a module without restart
by Stagg Shelton 04 Feb '11

04 Feb '11

Is it possible to reload a module such as siptrace without requiring a full restart of kamailio. I have changed my db_url modparam for the siptrace module and would like to avoid restarting kamailio if possible. Thanks Stagg

3 3

Problem with ser
by Asuncion Merino Rodriguez 02 Feb '11

02 Feb '11

Hello, i have insalled a ser server sucessfully. To try it, we was registered by a softphone, but when i am calling to the another registered user, he hang the phone but we cant ear anything. Can somebody help me? Thanks

1 0

Redundancy between 2 Kamailio servers
by Danny Dias 02 Feb '11

02 Feb '11

Hello my friends, I wonder if someone has done this before? i would like to implement 2 Kamailio servers with redundancy, something like the following: SERVER_A is working as the primary sip proxy (virtual IP as the sip signaling), if it fails, the other server (sleeping) should UP the virtual IP and takes all the traffic....i guess there is many Linux implementations thah can do this, but i would like to know if there is someone that has done this before and hear some recomendations... I've read some difficulty in the synchronisation of registrations because Kamailio works best when it stores registrations in memory and registrations are constantly changing - they expire and are renewed, as well as new ones joining and old ones leaving. To make the failover solution function seamlessly, it is necessary to synchronise the in-memory registrations between the primary and the backup server . This can be done by forking a copy of the registration request to the backup server, but there are some practical problems in doing this, has anyone do something with this? Thanks in advance!

6 21

Weird IPv6 Registration Issue
by Jon Farmer 02 Feb '11

02 Feb '11

Hi I am experimenting with a dual homed (IPv4 and IPv6) Openser 1.1.0-tls server. If I register using a IPv6 address then the initial registration works and the details are saved in the location table. However when the UA trys to re-register on expiry the SIP returns a 200 OK but the location data is not updated in the table. Looking at the syslog it is reporting that it didn't save the data as there was an existing entry on index with key 1. This is referring I presume to the primary index on "username, domain, contact) on the location table. So what happens is the first location entry expires and gets deleted from the table and thus there is no record for the subsequent register and thus openser thinks the UA has dropped off until the next re-register when because there is no longer a location record the data gets saved correctly. This doesn't happen if I register the same UA on IPv4 on the same openser server. Anyone got any ideas why this is happening? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 2

ACC error | failed to insert into database
by alex pappas 02 Feb '11

02 Feb '11

Hi all, Hve anyone seen before the following error? *Kamailio acc [acc.c:398]: failed to insert into database* After a Kamailio restart it is ok but it start again afetr x time. Thank you Alex

2 2

Re: [SR-Users] t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

argh. I just mixed up the true/false return values in the eMail but you still got it :-) Thanks a lot! ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 01 Feb 2011 20:31:17 +0100 Subject: Re: [SR-Users] t_local_replied() ? > Hello, > > not sure if you ask about the options, or you tried them and don't give > you the needed feature, since there are some improper true/false return > codes in your email. t_branch_replied() will return false if the 408 is > generated locally. > > Cheers, > Daniel > > On 2/1/11 8:25 PM, Bernhard Suttner wrote: > > Hi, > > > > just to be sure: > > > > - If the gateway does send back a 100 Trying and then a 408 is detected > within failure_route the method t_branch_replied does return false (means: > gateway is up and running) - dont go to next gateway (dispatcher) > > > > - If the gateway does not respond and a 408 is detected within > failure_route (= 408 was generated from kamailio) t_branch_replied does > return true (means: gateway is down) - go to the next gateway (dispatcher) > > > > Is that correct or am I wrong? > > > > Best regards, > > Bernhard > > > > > > > > ----- Original Message ----- > > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > > To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] > > Cc: sr-users(a)lists.sip-router.org > > Sent: Tue, 01 Feb 2011 13:30:54 +0100 > > Subject: Re: [SR-Users] t_local_replied() ? > > > > > >> Hello, > >> > >> On 2/1/11 12:08 PM, Bernhard Suttner wrote: > >>> Hi, > >>> > >>> I am using the dispatcher module and want to check within the > >> failure_route if the 408 was internally generated from kamailio or it was > >> received from the dispatcher gateway. > >>> There was previously a function called t_local_replied() in the > TM-module > >> but I could not find this function in the current documentation. Was it > >> removed? > >>> Is there a alternative to check if the 408 was local generated or if it > >> was received from the peer (= from the dispatcher gateway)? > >> see the example of: > >> http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied > >> > >> Cheers, > >> Daniel > >> > >> -- > >> Daniel-Constantin Mierla > >> http://www.asipto.com > >> > >> > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

1 0

Re: [SR-Users] t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

Hi, just to be sure: - If the gateway does send back a 100 Trying and then a 408 is detected within failure_route the method t_branch_replied does return false (means: gateway is up and running) - dont go to next gateway (dispatcher) - If the gateway does not respond and a 408 is detected within failure_route (= 408 was generated from kamailio) t_branch_replied does return true (means: gateway is down) - go to the next gateway (dispatcher) Is that correct or am I wrong? Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 01 Feb 2011 13:30:54 +0100 Subject: Re: [SR-Users] t_local_replied() ? > Hello, > > On 2/1/11 12:08 PM, Bernhard Suttner wrote: > > Hi, > > > > I am using the dispatcher module and want to check within the > failure_route if the 408 was internally generated from kamailio or it was > received from the dispatcher gateway. > > > > There was previously a function called t_local_replied() in the TM-module > but I could not find this function in the current documentation. Was it > removed? > > > > Is there a alternative to check if the 408 was local generated or if it > was received from the peer (= from the dispatcher gateway)? > see the example of: > http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 1

Radius and kamailio: avp unknown attribute
by Dominguez Jover, Ricardo 01 Feb '11

01 Feb '11

Hi everybody, I have a problem with Kamalio and Radius, when a try to authenticate a user I get the following: Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: rc_avpair_new: unknown attribute 5 Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: ERROR: auth_radius [sterman.c:412]: authorization failed In fact I can't see any packet sended to the Radius Server from Kamailio. I've read several posts and I've chmod of /var/run/radius.seq, include dictionary.sip in dictionary.radius and so on. Changing "if (!radius_www_authorize("$td"))" with " if (!radius_www_authorize(""))" I don't get any error but and can't neither see any packet sended to the Radius Server. I've checked the connection between radius client and radius server and it's OK, I get: Received response ID 91, code 2, length = 211 Any help would be appreciated. Kind regards, Ricardo Dominguez

2 2

Trunk connection asterisk-kamailio
by Mino Haluz 01 Feb '11

01 Feb '11

Hi, I have a question about kamailio-asterik interconnection. I'd like to connect 1000 numbers with a trunk, but it would be painful to add 1000 trunks on asterisk. Do you have some idea how could I group those numbers into one trunk connected to kamailio? asterisk and kamailio configuration settings. Thank you very much.

2 1

t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

Hi, I am using the dispatcher module and want to check within the failure_route if the 408 was internally generated from kamailio or it was received from the dispatcher gateway. There was previously a function called t_local_replied() in the TM-module but I could not find this function in the current documentation. Was it removed? Is there a alternative to check if the 408 was local generated or if it was received from the peer (= from the dispatcher gateway)? Thanks in advance! Best regards, Bernhard

2 1

kamailio and sip-router at FOSDEM 2011 - social event
by Henning Westerholt 31 Jan '11

31 Jan '11

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2011/) takes place in Brussels. As in the last two years 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 - Marius Zbihlei - Henning Westerholt This year we'll also be present in the Open Source Telephony room on sunday with two talks, "Scaling location services in large SIP networks with Kamailio from Marius and myself, and "Unifying SIP and Web worlds with Lua (Kamailio)" from Daniel. Olivier Taylor agreed helping us again in choosing a nice place for the evening. 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. Looking forward to see you in Brussels! Best regards, Henning P.S.: Photos of the 2009 event can be found on the web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Karlsruhe, Germany

2 3

xlogl and pseudo variables
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, I am using xlogl for debugging and logging the most important actions within the kamailio script. Does it take much resources if I use the pseudo variables with the xlogl statements? Example: xlogl("L_NOTICE", "Route $rm to destination! F=$fu T=$tu IP=$si \n"); I´m planning to do this in all the xlogl statements to be able to debug / "see whats going on" on the system. Thanks in advance. Best regard, Bernhard

2 1

Re: [SR-Users] onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, that was _really_ fast. Thanks. :-) One more question: Could I somehow test with the route[NAT] for example, if the request was from the main route { ... } or from the onreply_route[] ? Best regards, Bernhard ----- Original Message ----- From: Alex Balashov [mailto:abalashov@evaristesys.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org [mailto:sr-users@lists.sip-router.org] Sent: Sat, 29 Jan 2011 21:37:34 +0100 Subject: Re: [SR-Users] onreply_route and failure_route > Yes. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jan 29, 2011, at 3:35 PM, "Bernhard Suttner" <bernhard.suttner(a)winet.ch > > wrote: > > > Hi, > > > > is it possible to call a route within onreply_ and failure_route? I > > mean, within the main route { ... } its possible to call the route > > [NAT]. Is this also possible for this "special" routes? If this is > > possible, is it also allowed to call a route which will normally > > called from the main route { ... } > > > > Thanks in advance. > > > > Best regards, > > Bernhard > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > > list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, is it possible to call a route within onreply_ and failure_route? I mean, within the main route { ... } its possible to call the route[NAT]. Is this also possible for this "special" routes? If this is possible, is it also allowed to call a route which will normally called from the main route { ... } Thanks in advance. Best regards, Bernhard

2 1

Joining strings
by Lee Archer 29 Jan '11

29 Jan '11

Hi, is it possible to join strings? I'd like to prepend a number to a string prior to processing it to an int? Thanks Lee

2 2

Can you use a bigint with Kamailio?
by Lee Archer 29 Jan '11

29 Jan '11

Hi, I am trying to run dispatcher based on the inbound number called. This works fine for some numbers but not all. Is there a bigint string function as this would solve my problems? The numbers I am receiving are 14 digits long which s.int is returning as something completely different to what it should be. Thanks Lee

2 2

LCR next_gw() - certain GW, source from other IP address
by Graham Wooden 29 Jan '11

29 Jan '11

Hi there, I have a requirement now that for one of my GW¹s in my LCR, needs to be sourced from another IP address (two different rate-decks with the carrier). This ratedeck will always be at least 2nd inline ... I was thinking something along the sorts ... Some pseudo code: failure_route [2] { . . if (next_gw()) { if ( name = 'gw_name') { force_send_socket(udp:ip.address.of.certain.ip:5060); } else { # just dummy, as we will send out our default IP for everyone else. } t_on_failure("2"); route(19); } . . . } What variable will have the value that's in the gw_name column? I can't go by the IP address in $ru because the destination IP is the same between the two ratedecks. Overall, Is that the best way to handle this requirement? Thanks all, -graham

3 2

Re: [SR-Users] LCR next_gw() - certain GW, source from other IP address
by GP Wooden 29 Jan '11

29 Jan '11

Same destination IP for both rate decks, so I need to go by something else, like the gw_name. ----- Reply message ----- From: "Stagg Shelton" <stagg(a)vocalcloud.com> Date: Sat, Jan 29, 2011 8:27 am Subject: [SR-Users] LCR next_gw() - certain GW, source from other IP address To: <sr-users(a)lists.sip-router.org> Do the provider gateways have the same IP address for the different rate plans? If the gateways are different then the following may be relevant. if (next_gw()) { # prepare for lcr failover t_on_failure("2"); if($rd=="aaa.aaa.aaa.aaa") { xlog ("L_INFO", "This is a request for Some Rate Plan"); force_send_socket(udp:xxx.xxx.xxx.xxx:5060); } else { xlog ("L_INFO", "This is a request for another rate plan"); force_send_socket(udp:yyy.yyy.yyy.yyy:5060); }; route(2); } else { xlog("No gateways found!"); Stagg On 1/29/11 9:10 AM, Graham Wooden wrote: > Hi there, > > I have a requirement now that for one of my GW¹s in my LCR, needs to be > sourced from another IP address (two different rate-decks with the carrier). > This ratedeck will always be at least 2nd inline ... > > I was thinking something along the sorts ... Some pseudo code: > > failure_route [2] { > . > . > if (next_gw()) { > if ( name = 'gw_name') { > force_send_socket(udp:ip.address.of.certain.ip:5060); > } else { > # just dummy, as we will send out our default IP for everyone > else. > } > t_on_failure("2"); > route(19); > } > . > . > . > } > > What variable will have the value that's in the gw_name column? I can't go > by the IP address in $ru because the destination IP is the same between the > two ratedecks. > > Overall, Is that the best way to handle this requirement? > > Thanks all, > > -graham > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 4

Kamailio Devel Training, Barcelona - update
by Daniel-Constantin Mierla 28 Jan '11

28 Jan '11

Hello, a quick update about the next training in Barcelona, Feb 10-11, 2011, focusing on Kamailio development. We got a bigger room (for second time :-) ), so we increased the capacity to 28 seats. We are already 23 people in the class and several more joining the dinner Thursday evening, so there is going to be a great time. If you plan to come, register asap, there is no time to get a bigger room, so once we are full booked this time, we close the participants list. You can see more details at: http://www.kamailio.org/w/2011/01/kamailio-development-training-barcelona-f… Also, do not forget Fosdem next weekend, it is a bunch of us going there, two talks about Kamailio, come and say hi: http://www.kamailio.org/w/2011/01/social-networking-event-brussels-belgium/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio as a routing engine
by Gang Liu 28 Jan '11

28 Jan '11

Hello, I wrote a H.323/SIP IWF program before, it was based on B2BUA framework. Currently I am planning to use Kamailio as internal routing server and let IWF become a class 4 soft switch.That is Inbound ---- H.323/ SIP -- > ---- SIP routing request-----> Kamailio IWF/SBC | Outbound <--- H.323/SIP -- <--- SIP ----------------------------- Is it ok to store all voip provider's information at database of Kamailio and pass all to IWF at private SIP headers and let IWF as a outbound proxy? Is Kamailio as a SIP redirect server or it is better to stay at call signaling path until call session ended? It is great to share any information about this. regards, Gang

2 1

Load_gw failed
by Thiago Maluf 27 Jan '11

27 Jan '11

Hi Kamailio Users, I'm trying to use LCR module in Kamailio 3.1.1 and I', having some problems with this function (load_gws). I'm my kamailio.cfg , I have this route: # SEARCH DEST route[NUMBER] { xlog("L_ALERT", "NUMBER IS $rU \n"); if ($rU=~"^55211020[0-9]{4}$") { # IS MY USERS route(LDAP_NUM2USER); } else { # NOT IS MY USER if (!load_gws(1,$rU)) { xlog("L_ALERT", "LOAD LOCAL GW FAILED $rU \n"); route(DSER); return; } else { xlog("L_ALERT", "LOAD GW DEBUG $avp(i:709)\n"); } ; if (!next_gw()) { xlog("L_ALERT", "NEXT LOCAL GW FAILED\n"); sl_send_reply("503", "Service not available - No gateways"); return; }; setflag(29); route(RELAY); return; }; }; When my call arrive to load_gws, I have this problem: (2198) ERROR: <core> [action.c:1158]: runtime fixup failed for load_gws param 2 (2198) ERROR: <core> [action.c:1511]: run action error at: /etc/kamailio/kamailio.cfg:472 I already saw the Kamailio Code and this error doesn't make sense. Would somebody Help me with it? Very thanks, Thiago Maluf. -- ---------------------------------------------------------------- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malufrj(a)gmail.com

2 1

SIP Recorder
by Danny Dias 27 Jan '11

27 Jan '11

Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is: How can i handle sip conversations recording when all the calls are passing through a Proxy Server? I do understand that the media is always peer to peer and the signaling goes through the Proxy, but in this case the media not only has to pass between the peers because it must be recorded. How should i handle this?

7 19

Kamailio CFG file size ??
by Jijo 27 Jan '11

27 Jan '11

Hi All, is there any limitation on the size of kamailo.cfg file. When i tried to add few avp_subst, then the kamailio starts showing up the error out of package memory on receiving SIP message. If i remove those lines then everything works fine.. avp_subst is simple one. Our kamailio cfg file is quite big.. Please let me know how to debug this issue? Here is the error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_row.c:116]: no private memory left 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:293]: could not allocate row 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:258]: failed to convert row #1 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:75]: failed to convert rows 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_dbase.c:416]: error while converting result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_query.c:137]: error while storing result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: avpops [avpops_db.c:333]: cannot do the query 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: maxfwd [mf_funcs.c:62]: parsing MAX_FORWARD header failed! 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: rr [loose.c:98]: failed to parse headers 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: pv [pv_core.c:1761]: do action failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [msg_translator.c:2229]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: rr [loose.c:70]: failed to parse To header field 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1740]: do action failed) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1381]: error parsing headers 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [data_lump.c:318]: ERROR: del_lump: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: textops [textops.c:1081]: no memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [backLinkMode] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_shv.c:367]: no more pkg mem 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:1257]: osb_get_osv_current_binding(): no binding available 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <script>: R1 - binding IP=0 not available

2 3

NAT Problem in Asterisk Realtime + Kamailio config
by andrzej.ciupek＠asterisk.edu.pl 27 Jan '11

27 Jan '11

Hello I have started to use configuration like: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I works fine, but I have problem with dealing with NAT. kamctl ul show gives me: AOR:: andrzej_104 Contact:: sip:andrzej_104@10.5.0.94:14368;rinstance=9c2b256a13210aa7 Q= Expires:: 46 Callid:: ZDUwYzIzMDNiZTI3MjkyNmFmOWEyOTI2ODcyNjVmNDg. Cseq:: 60 User-agent:: eyeBeam release 1102q stamp 51814 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 5087 AOR:: andrzej_103 Contact:: sip:andrzej_103@10.5.0.165:5060 Q= Expires:: 926 Callid:: 11631436-5060-1(a)10.5.0.165 Cseq:: 2257 User-agent:: Grandstream HT-502 V1.2A 1.0.3.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 6879 So when I have incomming call from Asterisk to Kamailio, Kamailio start to send call to LAN IP. I have tried to use help from: http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… But I have never get: Received:: and Cflag:: 192 it is always Cflag:: 0. field in my kamctl ul show. In my configuration Asterisk is Callweaver and is on the other server. I use rtpproxy-1.2.1 and kamailio-3.0.1 and mofified config from link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Please help.

2 2

RLS question
by "Andrés S. García Ruiz" 27 Jan '11

27 Jan '11

Hi, I'm trying to create a RLS "rls-services" document with a contacts-list inside it. Kamailio states that "No xcap document found for uri:x". When creating a subscribe I must set the uri to the service uri. What I want to know is the sql query Kamailio does in order to get the contacts list for that uri. For example: XCAP document: "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" + "<rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"" + " xmlns:rl=\"urn:ietf:params:xml:ns:resource-lists\">" + "<service uri=\"sip:restricted_areas_presentities@open-ims.test\">" + "<list name=\"friends\">" + "<rl:entry uri=\"sip:testuser01@open-ims.test\"/>" + "<rl:entry uri=\"sip:testuser02@open-ims.test\"/>" + "</list>" + "<packages>" + "<package>presence</package>" + "</packages>" + "</service>" + "</rls-services>"; Is stored in /xcap-root/rls-services/users/sip:restricted_areas@open-ims.test/index The subscribe is from sip:restricted_areas@open-ims.test to sip:restricted_areas_presentities@open-ims.test What am I doing wrong? Thanks, Andrés.

3 5

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