- sr-users - kamailio.org

Re: [SR-Users] Kamailio 5.1.6 Debian Package -> SEGFAULT
by Floimair Florian 08 Jan '19

08 Jan '19

This is the backtrace of 2 core-dumps i got just recently (both with the same timestamp). Nothing obvious that hits my eyes (at least no NULL pointer). Maybe you can see more in it. Backtrace #1: =========== [New LWP 99122] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007fca0d45e996 in t_should_relay_response (Trans=0x7fca067b2a60, new_code=401, branch=0, should_store=0x7ffff47edd44, should_relay=0x7ffff47edd40, cancel_data=0x7ffff47edf30, reply=0x7fca0d8fd020) at t_reply.c:1282 1282 t_reply.c: No such file or directory. (gdb) bt #0 0x00007fca0d45e996 in t_should_relay_response (Trans=0x7fca067b2a60, new_code=401, branch=0, should_store=0x7ffff47edd44, should_relay=0x7ffff47edd40, cancel_data=0x7ffff47edf30, reply=0x7fca0d8fd020) at t_reply.c:1282 #1 0x00007fca0d46317f in relay_reply (t=0x7fca067b2a60, p_msg=0x7fca0d8fd020, branch=0, msg_status=401, cancel_data=0x7ffff47edf30, do_put_on_wait=1) at t_reply.c:1786 #2 0x00007fca0d469154 in reply_received (p_msg=0x7fca0d8fd020) at t_reply.c:2537 #3 0x000055e77686fbe3 in do_forward_reply (msg=0x7fca0d8fd020, mode=0) at core/forward.c:747 #4 0x000055e776871a00 in forward_reply (msg=0x7fca0d8fd020) at core/forward.c:852 #5 0x000055e7768bd4e7 in receive_msg ( buf=0x55e776da0520 <buf> "SIP/2.0 401 Unauthorized\r\nVia: SIP/2.0/UDP 172.17.217.10;rport=5060;received=172.17.217.10;branch=z9hG4bKe4bf.c46a843afa3ff97f5b20df53926515a2.0;i=62\r\nVia: SIP/2.0/TCP 195.225.36.194:1797;rport=40047;"..., len=736, rcv_info=0x7ffff47ee4e0) at core/receive.c:364 #6 0x000055e7767ce4fe in udp_rcv_loop () at core/udp_server.c:554 #7 0x000055e77673a15d in main_loop () at main.c:1619 #8 0x000055e7767421fe in main (argc=13, argv=0x7ffff47eeb98) at main.c:2638 Backtrace #2: =========== [New LWP 99120] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007fca0d45e996 in t_should_relay_response (Trans=0x7fca067da3c0, new_code=200, branch=0, should_store=0x7ffff47edd44, should_relay=0x7ffff47edd40, cancel_data=0x7ffff47edf30, reply=0x7fca0d8fd020) at t_reply.c:1282 1282 t_reply.c: No such file or directory. (gdb) bt #0 0x00007fca0d45e996 in t_should_relay_response (Trans=0x7fca067da3c0, new_code=200, branch=0, should_store=0x7ffff47edd44, should_relay=0x7ffff47edd40, cancel_data=0x7ffff47edf30, reply=0x7fca0d8fd020) at t_reply.c:1282 #1 0x00007fca0d46317f in relay_reply (t=0x7fca067da3c0, p_msg=0x7fca0d8fd020, branch=0, msg_status=200, cancel_data=0x7ffff47edf30, do_put_on_wait=1) at t_reply.c:1786 #2 0x00007fca0d469154 in reply_received (p_msg=0x7fca0d8fd020) at t_reply.c:2537 #3 0x000055e77686fbe3 in do_forward_reply (msg=0x7fca0d8fd020, mode=0) at core/forward.c:747 #4 0x000055e776871a00 in forward_reply (msg=0x7fca0d8fd020) at core/forward.c:852 #5 0x000055e7768bd4e7 in receive_msg ( buf=0x55e776da0520 <buf> "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 172.17.217.10;rport=5060;received=172.17.217.10;branch=z9hG4bK3494.0e37cca5019fe3c95285a3464d0eaa5e.0;i=b3\r\nVia: SIP/2.0/TCP 192.168.0.102:53360;rport=53360;received=9"..., len=638, rcv_info=0x7ffff47ee4e0) at core/receive.c:364 #6 0x000055e7767ce4fe in udp_rcv_loop () at core/udp_server.c:554 #7 0x000055e77673a15d in main_loop () at main.c:1619 #8 0x000055e7767421fe in main (argc=13, argv=0x7ffff47eeb98) at main.c:2638

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nat_uac_test not recognized command in onreply_route
by Andrew Chen 07 Jan '19

07 Jan '19

Hey it's me again. So I'm following the instructions in the Kamailio 5.1.x wiki module page for websocket configuration and it specifically mention to set this block: onreply_route[WS_REPLY] { xlog("L_INFO", "[CSeq $cs] Is in WS_REPLY."); xlog("L_INFO", "[CSeq $cs] Sending $rs $rr to $sel(via[2].host) via $sel(via[2].transport) len: $ml"); xlog("L_INFO", "[CSeq $cs] \n$mb \n"); if(nat_uac_test("64")) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } When I did this and started up Kamailio, it gave me an error: Jan 3 22:19:43 sjomainkama55 kamailio: ERROR: <core> [core/cfg.y:3309]: yyparse(): cfg. parser: failed to find command nat_uac_test (params 1) I moved that if statement to a route block and I was able to start it up. The wiki page for nathelper module specify this function can be run in onreply_route as well: 5.5. nat_uac_test(flags) Tries to guess if client's request originated behind a nat. The parameter determines what heuristics is used. Meaning of the flags is as follows: - *1* - The “Contact” header field is searched for occurrence of RFC1918 or RFC6598 addresses. - *2* - the "received" test is used: address in the “Via” header is compared against source IP address of signaling - *4* - The Top Most “Via” is searched for occurrence of RFC1918 or RFC6598 addresses - *8* - The SDP is searched for occurrence of RFC1918 or RFC6598 addresses - *16* - Test if the source port is different from the port in the “Via” header - *32* - Test if the source IP address of signaling is a RFC1918 or RFC6598 address - *64* - Test if the source connection of signaling is a WebSocket - *128* - Test if the “Contact” header URI port differs from the source port of the request (Warning: this is might be legal or even intended combination in non NATted scenarios) All flags can be bitwise combined, the test returns true if any of the tests identified a NAT. This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE. is this some known bug in 5.1.x? Thanks -- Andy Chen achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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Unable to start Kamailio with acc_json
by Joonas Kamailio 07 Jan '19

07 Jan '19

Dear List, I have a problem with initializing acc_json. As soon as I try to load acc_json module, Kamailio fails to start. I get the following messages to log with debug 4 (non-related messages omitted): Jan 7 16:17:42 labkam2 kamailio[51089]: INFO: acc_json [acc_json_mod.c:99]: mod_init(): janson version : 2.9 Jan 7 16:17:43 labkam2 kamailio: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe Running 'bt full' on coredump tells following: (gdb) bt full #0 strlen () at ../sysdeps/x86_64/strlen.S:106 No locals. #1 0x00007f086d0a69fd in mod_init () at acc_json_mod.c:120 __func__ = "mod_init" #2 0x000055f1699cc6a2 in init_mod (m=0x7f087c0983d0) at core/sr_module.c:897 __func__ = "init_mod" #3 0x000055f1699cca16 in init_modules () at core/sr_module.c:926 t = 0x29ceb62d i = -1 __func__ = "init_modules" #4 0x000055f16989445a in main (argc=3, argv=0x7ffc56d94378) at main.c:2617 cfg_stream = 0x55f16beb0010 c = -1 r = 0 tmp = 0x7f087c9512e7 "__vdso_getcpu" tmp_len = 32764 port = 1457078704 proto = 32520 options = 0x55f169c019f0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 2201341167 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0xf0b0ff <error: Cannot access memory at address 0xf0b0ff> st = {st_dev = 19, st_ino = 306346, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 80, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1543318847, tv_nsec = 544329773}, st_mtim = {tv_sec = 1546870639, tv_nsec = 902332400}, st_ctim = {tv_sec = 1546870639, tv_nsec = 902332400}, __glibc_reserved = {0, 0, 0}} __func__ = "main" I have loaded "acc" module before loading "acc_json", are there any other mandatory prerequisites? I am trying to use acc_json to output accounting data to syslog. I just recompiled latest 5.2 kamailio but problem still occurs. Kamailio version below: root@labkam2:/opt/git/kamailio# kamailio -v version: kamailio 5.2.0 (x86_64/linux) c5216b flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c5216b compiled on 16:13:00 Jan 7 2019 with gcc 6.3.0 Best Regards, Joonas Keskitalo

2 2

Dispatcher probing
by Duarte Rocha 07 Jan '19

07 Jan '19

Hey. I have another question regarding the Dispatcher Probing. In my SBC, a Gateway is defined by it's IP, Port and SAP (Kamailio's IP used for signaling). Those 3 parameters are defined in the Gateway table. In this case ( using Options for probing), i only have access to the destination IP ($dd) and Port ($dp). I need to know the SAP associated to the gateway that is currently being probed, in order to force the correct socket in the OPTIONS request. Is there any way to get more info about the db line that is being read by dispatcher for the probing? If that's not possible i was thinking about querying my db for the Ip:Port pair and if i got more than one result, do forking using the the SAPs in the results, sending multiple OPTIONS through the multiple sockets. In this case, if only one of the OPTIONS is answered, will the gateway still appear as "Active" ? Best Regards, Duarte Rocha

3 2

Kami, rtpproxy, T38 fax - anyone?
by yu＠yu-boot.ru 07 Jan '19

07 Jan '19

Yet again, fighting against strange fax behavior on Kamailio. When call is routed from device behind Kami and uplink is plain old Mediant2000 gateway to E1, T.38 always correctly established via reinvite and fax works perfectly. When call is routed to anything else including local Asterisk or Freeswitch, seems called don't hear fax noises and quickly throw BYE to me. Asterisk just don't see anything about UDPTL. Even more confusing, "near" device drops a call when detects fax tones from Asterisk or Yate. What can it be? SDP seem to be all transformed as they should be, I cant see anything wrong or different on SIP dumps. Rtpproxy is on bridge mode. Local device is Audiocodes MP118 and on private addressing, another end of Kami is on static IP. Voice calls work perfectly on this configuration. I can provide anything configs dumps etc, if it will help to solve this issue. Anyone using rtpproxy bridge and working T.38? Help me please. -- sent from myMail for Android

2 1

Sipt redirection number
by Duarte Rocha 04 Jan '19

04 Jan '19

Greetings, Regarding the module sipt and $sipt(redirection_number). Does this PV can show non digits correctly? I have a redirection number that has one char in the middle, like this "12C34567899". However, when i print the PV, i get "12<34567899". How can i fix this? Thanks, Best Regards, Duarte Rocha

2 1

Best place to change mem settings
by Duarte Rocha 03 Jan '19

03 Jan '19

Greetings, As far as i can tell, i have 3 ways to permanently change the memory settings in Kamailio : /etc/init.d/kamailio , /etc/default/kamailio and src/core/config.h. What's the priority between them? If i have different values in all 3 of them, which one will be used? Which is one is the best to use reliably? I recently upgraded my Kamailio from 5.0 to 5.2 and i need 12Mb of PKG. Before starting the new version for the first time i changed the values in /etc/default and the memory error didn't go away. It only worked after i changed it in core/config.h and recompilled. After that, i started Kamailio and i could do the changes in /etc/default without any problem. Does this happen for a reason? Best Regards, Duarte Rocha

3 2

logging destination IP and proto on reponse
by Andrew Chen 03 Jan '19

03 Jan '19

Hey all, Any way to log destination IP and proto of SIP responses? I'd tried to use majority of the pseudovars, but they all just return <null>. Thanks. -- Andy Chen achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

5 8

rtpengine in stateless kamailio
by Yuriy Gorlichenko 03 Jan '19

03 Jan '19

Hi! Happy new year to all!!! Look like I am first in this year wit hthe questions in this list :-). I'm using stateless kamailio and RTPengnine to build some kind of the stateless cluster I found that kamailio keeps some data in the SHMEM in case of using RTPengine module even if it is not a rtpengine_manage function but offer/answer/delete In this case if INVITE (offer) handled by 1-st kamailio in my cluster, and BYE/CANCEL handled by second kamailio in the cluster - 1-st kamailio (which has been used for offer) will hase kinda internal "memory leak" (in SHMEM it never decrased) At the rtpengine module source I found some transation dependencies for the rtpengine_manage function but did not find for the offer/answer/delete I supposed these 3 functions just sending requests to the rtpnegine with keys and not storing anything So my question - is it possible to use RTPengine module in stateless way to avoid internal "memory leak" because in my scenario I have big chance never receive BYE/CANCEL on the machine that started handle particular call

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Re: [SR-Users] sr-users Digest, Vol 163, Issue 31
by Geovanny de Jesús García Rojas 03 Jan '19

03 Jan '19

estoy haciendo un trabajo con websockts y kamailio. Me gustaría saber como puedo vincular mi sevidor Kamailio instalado en una maquina virtual en el sistema operativo Ubuntu, como podria vincularlo con los websockets creados , me gustaria saber como es ese proceso. no tengo mucho conocimiento de ello.. espero su respuesta gracias El lun., 31 de dic. de 2018 a la(s) 06:00, < sr-users-request(a)lists.kamailio.org> escribió: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Phonenum Module (Soltanici Ilie) > 2. Re: Phonenum Module (Joel Serrano) > 3. Re: Phonenum Module (Joel Serrano) > 4. Re: Phonenum Module (Soltanici Ilie) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 30 Dec 2018 21:21:36 +0300 > From: Soltanici Ilie <iliesh(a)mail.ru> > To: sr-users <sr-users(a)lists.kamailio.org> > Subject: [SR-Users] Phonenum Module > Message-ID: <1546194096.91991005(a)f406.i.mail.ru> > Content-Type: text/plain; charset="utf-8" > > Hi All, > > How can i get Local Number by using phonenum module? > What is the variable for that? > > Thank You >

2 1

Phonenum Module
by Soltanici Ilie 30 Dec '18

30 Dec '18

Hi All, How can i get Local Number by using phonenum module? What is the variable for that? Thank You

2 3

Logging from cfg file
by Muhammad Allaudin 29 Dec '18

29 Dec '18

Hi everyone, I have recently started working on Kamailio, I am trying to learn the basics. For "Register" route, I am trying to print the log but log is not appearing in log file. Other logs are there, only this one is missing. *if(is_method("REGISTER")){* * log(2, "Register request");* *}* I will be really helpful if you can help me with this. Thanks & Kind Regards *Regards,* *M.Allaudin* *Software EngineerNowtel GroupPhone # +92-333-8291874Blogger @ allaudin.github.io <http://allaudin.github.io>*

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A bit trick in msg_translator.c source file
by Mojtaba 29 Dec '18

29 Dec '18

Hello , Does anybody know why the lenght of both ip-address (s and name) in check_via_address function in msg_translator.c source file dose not check first? Actually i think it's better do. For example if via:192.168.122.1 and src:192.168.122.100, the result returned true, as they are not the same! With.Regards. Mojtaba Happy new year --Mojtaba Esfandiari.S

2 1

Re: [SR-Users] kamailio does not responde if an rtpengine is unreachable
by David Escartin 29 Dec '18

29 Dec '18

hello Daniel Thanks a lot for the update. We will also test it. It has not 100% relation with this issue, but i only wanted to share the setup we have for cases where a rtpengine fails having high traffic load, to minimize the impact on the kamailio processes. modparam("rtpengine", "queried_nodes_limit", 2) modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "rtpengine_tout_ms", 350) considering we don't use sets with more than 2 rtpengine instances, at least for retry attempts. And your rtpengine instances are in the same network too. this works quite fine for us, there are some few secs of impact while the rtpengine is marked as disabled, but the system recovers quite ok. best regards david -----Original Message----- From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> Sent: Friday, December 28, 2018 9:15am To: "Juha Heinanen" <jh(a)tutpro.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] kamailio does not responde if an rtpengine is unreachable I just pushed a series of commits trying to rework how loading (and reloading) of rtpegines list is done, to avoid that sync'ed probing, which can take long if any of the rtpengines is down. Now, building the local (per process) structures/sockets for rtpengines during kamailio start up is done without locking. This is guarded by the fact a reload command can be executed only after all children were initialized (added also with these commits). Moreover, the probing of rtpeningesis done only by child process 1, because the status is stored in shared memory list, so it is visible in all children. Based on my understanding there, doing probing from all processes is useless now, that was probably kept from the time when the list was not stored in shared memory, from the early rtpproxy times. There is also a restriction on how often the rtpengine list can be reloaded, now having a 10 seconds interval guard. I added this because the reload is done over the old list, not building a new list to swap with the old one. So it requires some time to walk through the existing list and update based on the new records. I went this way for now, even building a new list may be better/safer in long term, but it would require more work. I also wanted to avoid being very intrusive right now, given that those patches would need to be backported. The last relevant change was to use a version number to discover when a reload was done. So far, as I understood, it was relying on the number of rtpengines, but one may trigger a reload with same rtpengines, but different attributes (e.g., disabled or not). Having a version number is better in detecting when each worker needs to rebuild its local list of sockets, as well as for troubleshooting, because a value is increased with each reload, so easier to track if it was done or now. I didn't have time for any tests, so it would be good if you can test and report if works as expected. All related commits are in master, if they prove to work fine, we can backport all those patches. Cheers, Daniel On 26.12.18 12:46, Juha Heinanen wrote: > Daniel-Constantin Mierla writes: > >> I pushed a quick fix for the case when db support is not enabled, >> because these locks are useless in that case, so all children will do >> the rtpengine init at the same time, without waiting for the others: > Still took in rtpengine db mode about 2 minutes before kamailio became > responsive after start. > > -- Juha -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

kamailio does not responde if an rtpengine is unreachable
by Juha Heinanen 28 Dec '18

28 Dec '18

I noticed that if one rtpengine in a set is unreachable, kamailio 5.2 does start, but does not process any SIP requests. Is this intentional? -- Juha

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Re: [SR-Users] Kamailio 5.2 0IMS configuration documentation
by Woscek, Martin W. 27 Dec '18

27 Dec '18

Hi Yasin, That VM was built with 4.3.4. It did help in fixing the scripts to build the pcsf, scscf and icscf databases. The config files are different and being that I need to understand the configuration and getting it up and running, isn’t helping much more. Being I need an IMS for lab testing, perhaps 4.3.4 does what I need, then maybe 5.2.0 is not required. Thanks, MArtin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of YAS0 CANER Sent: Thursday, December 27, 2018 12:56 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] Kamailio 5.2 0IMS configuration documentation Hello i found it. https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ Good luck Yasin caner YAS0 CANER <caner_yaso(a)hotmail.com<mailto:caner_yaso@hotmail.com>> şunları yazdı (27 Ara 2018 20:50): Hello, ims vmware snapshot was shared by Franz Edler in mailling list. it helps you to build it. Good luck iPhone’umdan gönderildi Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> şunları yazdı (27 Ara 2018 19:53): Hi all, Ive been in the process of standing up an instance of Kamailio 5.20, in particular the IMS. The P-CSCF, I-CSCF and P-CSCF as well as the mysql are on this single VM. I’ve been slowly working thru the errors as I bring up the P-SCSF first. Can someone point me to detailed documentation for setting this up? What is the base minimum for basic IMS functionality (eg: do I need the mysql database)? Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio 5.2 0IMS configuration documentation
by Woscek, Martin W. 27 Dec '18

27 Dec '18

Hi all, Ive been in the process of standing up an instance of Kamailio 5.20, in particular the IMS. The P-CSCF, I-CSCF and P-CSCF as well as the mysql are on this single VM. I've been slowly working thru the errors as I bring up the P-SCSF first. Can someone point me to detailed documentation for setting this up? What is the base minimum for basic IMS functionality (eg: do I need the mysql database)? Thanks, Martin

2 2

Dispatcher Probing
by Duarte Rocha 27 Dec '18

27 Dec '18

Greetings, I have Kamailio with the Dispatcher module and it's probing mechanics set up. I want to be able to access the OPTIONS request generated by Kamailio in the request_route or somewhere else in the configuration file in order to modify it. Is that possible? Best Regards, Duarte Rocha

3 4

xlog $mb produces ERROR
by Andrew Chen 27 Dec '18

27 Dec '18

Hey community, I'm currently trying to config our new 5.1.2 kamailio server and I want to xlog $mb. Whenever I do this, I get an ERROR logged to my syslog Dec 26 21:15:58 sjomainkama55 /usr/sbin/kamailio[5954]: {1 1 INVITE <padding>@<padding>} ERROR: <script>: INVITE sip:<padding>@ sjomainkama55.a.fuzemeeting.com:443 Does anyone know why this is? Also, I noticed I can't get rid of #012#015 at all. I modified the rsyslogd.conf file as instructed here: "i need to add the next line to /etc/rsyslog.conf "$EscapeControlCharactersOnReceive off" and when I did this, the control characters disappear but all it printed out was the first line of the SIP message. I also try to assign $mb to a $var() and do a {re.replace} on that string and that didn't work either. I ended up having to create a side script to parse the whole /var/log/syslog and do regex replace the #012#015 with real newline which is not what I really want for me or anyone else who needs to analyze the Kamailio logs. Please advise. Thanks. -- Andy Chen achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 4

Re: [SR-Users] [EXT] Re: Using a network STUN Server
by Wilkins, Steve 27 Dec '18

27 Dec '18

Thank you, I just was not sure what else would cause the relayed packets to not be sent out to my fios router. As mentioned, I can pick any other server in my network and I can see, in the pcap file, that the relay is attempted to the selected server. I verified our ACL and it is it open for TCP output to any IP and Port so that is not the issue. -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Alex Balashov Sent: Wednesday, December 26, 2018 12:55 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] Using a network STUN Server As STUN is a client-side construct, I am not aware of any functionality Kamailio might have to make *use* of a STUN server for its own representations about its reachability. But I could be wrong. -- Sent from mobile. Apologies for brevity and errors. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

Fwd: Dialog getting disconnected as soon as call is ended
by Subham Singhal 27 Dec '18

27 Dec '18

---------- Forwarded message --------- From: Subham Singhal <subham.singhal(a)zemosolabs.com> Date: Sat, Dec 8, 2018 at 10:19 AM Subject: Re: [SR-Users] Dialog getting disconnected as soon as call is ended To: <miconda(a)gmail.com> Hello, I am getting this error in kamailio console. I thought websocket connection is getting disconnected before receiving bye message so I tried to keep tcp connection open. But that also didn't work. On Fri 7 Dec, 2018, 6:48 PM Daniel-Constantin Mierla <miconda(a)gmail.com wrote: > Hello, > > which application is printing that log message? > > Cheers, > Daniel > On 03.12.18 08:21, Subham Singhal wrote: > > Hello, > > I am new to kamailio. I have been trying to integrate kamailio with > asterisk. I am able to send INVITE and get RTP packets as well, but I am > unable to receive BYE from asterisk. Call is getting terminated through > following websocket error. > "WebSocket connection to 'wss://172.16.16.169:5061/' failed: One or more > reserved bits are on: reserved1 = 1, reserved2 = 0, reserved3 = 0." > > -- > Thanks and Regards > Subham Singhal > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference -- www.kamailioworld.com > Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com > > -- Thanks and Regards Subham Singhal

1 0

Using a network STUN Server
by Wilkins, Steve 26 Dec '18

26 Dec '18

Hello All, Can I point Kamilio to use a STUN server located on my network? Kamilio will not relay to my fios router but will relay to other Servers on my network. Thank you,

2 1

stateless error reply used: transaction canceled
by Евгений Голей 26 Dec '18

26 Dec '18

Hello! I would be grateful if you could tell me what the causes of the error in Kamailio version 5.1.5. 2018-12-26T20: 28: 18.3875 "ERROR: tm [tm.c: 1545]: _w_t_relay_to (): t_forward_noack failed" 2018-12-26T20: 28: 18.3879 "ERROR: sl [sl_funcs.c: 362]: sl_reply_error (): stateless error reply used: transaction canceled (487 / SL)" B.R. Evgeniy

1 0

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 24 Dec '18

24 Dec '18

With the season holiday just starting, it's again a good time to express my thanks and greetings to all the friends, developers and community members that made 2018 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 - www.kamailioworld.com

2 1

app_python3 and kamailio 5.2.0 docker
by Grant Bagdasarian 24 Dec '18

24 Dec '18

Hello, I've pulled the Kamailio 5.2.0 Debian Stretch Docker image, and tried to load app_python3, but without success. It's also not in the Debian repository, at least not part of the kamailio-python-modules package. What is the current, recommended, way of installing the app_python3 module? Source files only? Regards, Grant

2 2

failure route is not executed when suspended invite is cancelled
by Koray Vatansever 24 Dec '18

24 Dec '18

Hi all, I'm trying to upgrade kamailio from 4.2.7 to 5.1.5 and I realized that when CANCEL is received for a suspended INVITE, failure route is not executed in release 5.1.5. In 4.2.7 failure route is executed for this scenario. Is this the expected behaviour or am I missing something?

2 7

Manipulate sip header on an in-dialog negative reply
by Olli Attila 24 Dec '18

24 Dec '18

Hello, We have a call case where our softswitch replies to in-dialog re-invite with "SIP 491 Request pending". This happens when a customer sip device is trying to re-invite the session too fast even though the softswitch is still processing the earlier request. Kamailio operates between this customer device and the softswitch. The softswitch is adding a P-Charging-Vector header to this SIP 491 reply which we want to drop (remove_hf...) from the reply when we route the SIP 491 message back to the customers device. I tried to drop this inside in-dialog failure route but it seems that Kamailio forwards the orginal 491 message statelessly to customer and the header still extists there eventhough failure route has deleted it. I guess this modification should be done in a sateful manner for Kamailio to actually write the changes to the outgoing reply towards the customer device. Any suggestions how the 491 reply could be edited and then forwarded onwards to the customer device? Cheers, --Olli

4 5

Large PDD
by Zhan Bazarov 22 Dec '18

22 Dec '18

Hello, I have kamailio+asterisk platform. Everything works fine. But, sometimes, for unclear reasons, the PDD(8000 and more) rises to an incoming call(at the stage when the call goes to an asterisk). I can not figure it out myself (SIP is clean, hardware metrics are ok) and it happens in random order (under load and without it). Anybody have ideas, what in can the reason?

2 1

Build Kamialio Docker Image Failed.
by Zhong, Alec (NSB - CN/Qingdao) 21 Dec '18

21 Dec '18

Hello, When we build kamailio image, the below error happened. At the same time, we can not access http://security.debian.org/debian-security/dists/jessie/updates/main/binary… Look like the Packages had been moved. Could you please help check below issue? Step 1/21 : FROM debian:jessie ---> 4eb8376dc2a3 Step 2/21 : MAINTAINER L. Mangani <lorenzo.mangani(a)gmail.com> ---> Running in ebf133dd0924 ---> 0abb8a3fcf82 Removing intermediate container ebf133dd0924 Step 3/21 : ENV HOME /root ---> Running in a92f7359fefd ---> e9872a1d4892 Removing intermediate container a92f7359fefd Step 4/21 : ENV DEBIAN_FRONTEND noninteractive ---> Running in eab86e90f492 ---> 25b976249d19 Removing intermediate container eab86e90f492 Step 5/21 : RUN apt-get update -qq ---> Running in e1132468ff42 W: Failed to fetch http://security.debian.org/debian-security/dists/jessie/updates/main/binary… 503 Service Unavailable E: Some index files failed to download. They have been ignored, or old ones used instead. The command '/bin/sh -c apt-get update -qq' returned a non-zero code: 100 make: *** [build] Error 100 Best Regards, Alec Zhong

3 3

Kamailio as an SBC - five years on
by Alex Balashov 20 Dec '18

20 Dec '18

Hi John, Sorry, I accidentally deleted your original e-mail and thus am breaking threading; my apologies for that. As a matter of fact, I wrote an article recently to address the very questions you ask: http://www.evaristesys.com/blog/kamailio-as-an-sbc-five-years-on/ As always, it is a matter of professional opinion, and I do not speak for the project nor the intentions of the core developers. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 2

Rather noob question about functionality of Kamailio as an SBC
by John Millican 20 Dec '18

20 Dec '18

Hello, I have read the article at: https://likewise.am/2013/03/10/kamailio-as-an-sbc-session-border-controller/ Since this was written back in 2013, I was wondering if there has been any changes/additions/improvements in Kamailio since then that would have an impact on the conclusions made in the above article? Where I work we are in the process of 1) determining what functionality we need now and in the not too distant future 2) based on the choices made in #1, do we need a fully functional "professional" SBC or can we get away with a "pedestrian" SBC and use an application like Kamailio. Generalizations are ok of course but pointers to specifics would be awesome if possible. I know this list is not here to do my homework for me. I will be doing more research myself but I am sure there are many folks here that have gone down this road already and am hoping to benefit form their experience while I build my own knowledge. Thanks, John

1 0

Stateless non200 ACK processing
by Serge S. Yuriev 20 Dec '18

20 Dec '18

Hello, I'm trying to build pretty simple stateless proxy to stay between gw and softswitch. Most working properly but not for non200 ACKs. For some reason softswitch not accepting my ACK. Trace (note different branch in top Via for INVITE and ACK towards switch) https://pastebin.freeswitch.org/view/8f1433b0 Simplified script https://pastebin.com/u9DTSqcS I'm pretty new to this so believe doing something wrong. -- Serge S. Yuriev Senior VoIP engineer

2 4

TOPOS with Redis Issues
by Aleksandar Sosic 20 Dec '18

20 Dec '18

Hi Guys, We're having issues trying to use redis as the db for topos. The conf is the following: ``` ... loadmodule "ndb_redis.so" loadmodule "topos.so" loadmodule "topos_redis.so" ... modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=1") modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 12600) modparam("topos_redis", "serverid", "srv1") ... ``` We've tried also with a redis server running on another machine with no success. The errors I've got in the kamailio logs are the following: ``` 2(22) ERROR: dialog [dlg_handlers.c:212]: populate_leg_info(): empty CSeq number 2(22) ERROR: dialog [dlg_handlers.c:506]: dlg_onreply(): could not add further info to the dialog ``` and ``` 17(37) ERROR: dialog [dlg_req_within.c:117]: build_dlg_t(): invalid cseq 17(37) ERROR: dialog [dlg_req_within.c:383]: send_bye(): failed to create dlg_t 17(37) ERROR: dialog [dlg_req_within.c:87]: build_dlg_t(): no contact available 17(37) ERROR: dialog [dlg_req_within.c:383]: send_bye(): failed to create dlg_t ``` Kamailio version used is this: 5.2.0+0~20181218003543.18+xenial Thanks in advance for your time, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

2 3

Default memory
by Duarte Rocha 20 Dec '18

20 Dec '18

Greetings, How can i ibcrease permanently the Kamailio default private and shared memory size? Thanks in advance, Duarte Rocha

4 3

Memory allocation
by Duarte Rocha 19 Dec '18

19 Dec '18

Greetings, I have some things that i want to udnerstand about Kamailio's memory allocation. My settings are 1024Mb for shared memory and 12Mb for private memory. I'm running Kamailio on a machine with CentOS7 and 2Gb of RAM. As i understand, at startup Kamailio will allocate 1024+12*number_of_processes megabytes of memory. Is this correct? When i set up Kamailio to run with 150 processes, after starting, i still have 500Mb of free RAM and therefore not allocated to any process. That goes against what i wrote earlier. What is the best way to plan for memory specs in my machines according to the desired number of processes and shared\private memory defined? Best Regards, Duarte Rocha

2 1

reg_fetch_contacts always returns -1
by Denys Pozniak 19 Dec '18

19 Dec '18

Hello! I want to get some parameters from user location database via function reg_fetch_contacts(). But it returns -1 and $(ulc(caller=>aor)) is <null> respectively. Please help me to understand where issue is. Script: ... $var(c_uri)="sip:+1234567890@10.10.10.10:5065"; reg_fetch_contacts("location","$var(c_uri)","caller"); $var(rc) = $rc; xlog("L_WARNING","[LOCATION] caller=>aor: rc=$var(rc) aor=$(ulc(caller=>aor)) \n"); ... [root@kamailio]# kamcmd ul.lookup location customer1 { AoR: customer1 Contacts: { Contact: { Address: sip:+1234567890@10.10.10.10:5065 Expires: 861 Q: -1.000000 Call-ID: 00DB5F27-0B02-E966-92C3-A231C378F993(a)10.10.10.10 CSeq: 1787 User-Agent: SIPPER for PhonerLite Received: sip:2.2.2.2:15043 Path: [not set] State: CS_SYNC Flags: 0 CFlags: 64 Socket: udp:3.3.3.3:5060 Methods: 8095 Ruid: uloc-1-5c1a3512-66a6-1 Instance: <urn:uuid:00FB3C49-24E8-E811-ACAA-D72FCD766436> Reg-Id: 0 Server-Id: 1 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1545232794 Last-Modified: 1545232794 } } } [root@kamailio]# kamailio -v version: kamailio 5.1.6 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 13:53:57 Nov 12 2018 with gcc 4.8.5 -- BR, Denys Pozniak

1 2

RFC: updates to some core functions
by Daniel-Constantin Mierla 19 Dec '18

19 Dec '18

Hello, it was brought into discussions several times in the past about core functions not accepting variables in the parameters. I think it is time to update them during the 5.3 release development. For few of them, I added in the past some alternative function in the corex module (e.g., force_send_socket() in core and set_send_socket() in corex module). So, I see two options: 1) add a function with similar name in corex module and same behaviour like the one from core 2) remove the function export from the core and export one with the same name from the corex module First one will ensure that configs using the functions right now keep working without any update. The second one will be better in long term from the point of documentation (no duplicated docs), but there might be few cases that would require updates in the config -- iirc, there are some functions that can get special tokens in the parameters (like forward(uri:host, uri:port)), they will get an equivalent with variables, but old config will not be compatible. Obviously the reason for this email is to ask the developers and users what would be the preferred way from own point of view. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

7 11

passing $avp to t_relay functions.
by YAS0 CANER 19 Dec '18

19 Dec '18

Hello , is there a way to pass $avp values to t_relay function. i tried ; t_relay t_relay_udp/tcp t_relay_to t_relay_avp functions log 0(75155) ERROR: tm [tm.c:2456]: fixup_t_relay_to(): fixing ($avp(dd), 1) 0(75155) ERROR: tm [tm.c:2459]: fixup_t_relay_to(): param type 8 0(75155) ERROR: tm [tm.c:2456]: fixup_t_relay_to(): fixing ($avp(flags), 2) 0(75155) ERROR: tm [tm.c:2459]: fixup_t_relay_to(): param type 11 0(75155) CRITICAL: tm [tm.c:2549]: fixup_t_relay_to(): invalid flags <$avp(flags)> 0(75155) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:489 0(75155) ERROR: <core> [core/rvalue.c:3860]: fix_rval_expr(): failure in cfg at line: 489 col: 42 0(75155) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:494 0(75155) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:495 in addition i have a look code , in fixup functions try to resolve host values so parameteres cannot find $avp . in conclusion , does kamailio need new function to pass $avp value? Can i try to code? Thanks.

2 6

Kazoo module problem.
by Володимир Іванець 18 Dec '18

18 Dec '18

Hello! I'm trying to test kazoo module but unfortunately it is not loading: *0(25107) ERROR: <core> [core/sr_module.c:498]: load_module(): could not open module </usr/lib64/kamailio/modules/kazoo.so>: /usr/lib64/kamailio/modules/kazoo.so: undefined symbol: event_set* It was built today together with other modules and Kamailio itself today from branch 5.2 on CentOS 6. librabbitmq version is 0.5.2-1. Does anyone has it running and if so what version is it? Thanks a lot!

3 4

Access to tables like location, imc_rooms and imc_members when kamailio is running
by hamidreza ramezani 18 Dec '18

18 Dec '18

Hello everyone I have a question regarding kamailio database, I see when kamailio is stopped, usrloc -> location and imc -> imc_rooms module tables shows some information. but when kamailio is running it's not. is it possible to manually insert to these tables, I don't want to use IMC commands like #join #accept ... is it possible to access location table and see which users are registered when kamailio is running? Best regards Hamid

2 1

Routing of Audio and Video Stream.
by Abhijit Dutta 18 Dec '18

18 Dec '18

Hi All, Is it possible to redirect only the audio stream towards Asterisk from Kamailio (Proxy) via RTPEngine and not to send the video stream to Asterisk. I am yet to get the complete reason why my client wants this to happen though. Thanks Abhijit

5 12

Re: [SR-Users] Setup dmq_usrloc
by Gertjan Wolzak 17 Dec '18

17 Dec '18

Hello Kamailions, Reading and understanding is a skill in it self.. Learned a bit more of that this morning. To be able to use the dmq_ursloc, just use: if(is_method("KDMQ")) { dmq_handle_message(); } So, I have registrar replication working, on with reading to get the rest working. With regards, Gertjan

2 1

Setup dmq_usrloc
by Gertjan Wolzak 17 Dec '18

17 Dec '18

Hello Kamailions, Im trying to setup registration replication with the use of the dmq and dmq_usrlog modules. I have configured two servers, configured kamailio on both. When I register a sip client on the one I get a nice KDMQ message on the second server, but the second server reacts with a 404 not found: U 2018/12/17 05:08:38.974495 10.10.0.1:6000 -> 10.10.0.2:6000 #3 KDMQ sip:usrloc@10.10.0.2:6000 SIP/2.0. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0. To: <sip:usrloc@10.10.0.2:6000>. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Content-Length: 360. User-Agent: kamailio (5.1.6 (x86_64/linux)). Max-Forwards: 1. Content-Type: application/json. . {"action":2,"aor":"test01(a)10.88.77.172","ruid":"uloc-5c177363-ec4-1","c":"sip:test01@10.88.77.140:59293;ob","received":"","path":"","callid":"zjucvmzUT2hNJCIfqeUXEsOaEzyHCs-4","user_agent":"CSipSimple_G3121-26/r2457","instance":"","expires":1545041786,"cseq":24435,"flags":0,"cflags":0,"q":-1,"last_modified":1545040886,"methods":8159,"reg_id":0,"server_id":0} U 2018/12/17 05:08:38.979062 10.10.0.2:6000 -> 10.10.0.1:6000 #4 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0;rport=6000. To: <sip:usrloc@10.10.0.2:6000>;tag=02f7a7b91eadf5e07797231cb2f64326-b794. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Server: kamailio (5.1.6 (x86_64/linux)). Content-Length: 0. . I think I have to configure a bit more than just the parameters of the dmq and dqm_usrloc modules. Ive googled, looked in the Sip routing with kamailio book, but no luck. Did not see any examples in the kamailio source, with regards to the dmq_usrloc module. I am not looking for direct answers, have not provided you with enough information for that, but I would appreciate information on how to correctly configure registration replication using the dmq/dmq_usrloc module... Thank you. With regards, Gertjan

1 0

Reminding again about private/direct emails from mailing lists discussions
by Daniel-Constantin Mierla 17 Dec '18

17 Dec '18

I see that still a lot of emails are sent directly to me after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon), I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended and there is practically 0 chance to get an answer from myself. The rule is also suggested in the mailing lists presentation page: - https://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get a reply instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

P-CSCF without Rx interface
by Tsvetan Filev 17 Dec '18

17 Dec '18

Hi All. May be this has been asked before but I was not able to find an answer. I setup IMS with 3 instances of kamailio: P-CSCF, I-CSCF and S-CSCF (I got the config files from here https://github.com/kamailio/kamailio/tree/master/misc/examples/ims). I would like to register two clients to the IMS and make a VoLTE like call. So I want to setup my P-CSCF to send SIP messages to my I-CSCF without Rx interface (without using PCRF). I'm not able to find how to tell P-CSCF where to direct the sip traffic. It obviously should happen in route[REGISTER]. Right now P-CSCF sends all REGISTER requests to itself and goes to a loop. Regards.

3 3

Sending to an announcement server from a failure_route
by Boden Garman 17 Dec '18

17 Dec '18

Hi list, I'm trying to play an announcement for my users when my PSTN gateway responds with a 404 Not Found (Reason: Q.850;cause=3), indicating a wrong number. I'm a bit lost and would appreciate any advice. What I am seeing is two parallel SIP invites, one to my announcement server (hooray!) and one to the PSTN gateway again (which I don't want). How do I stop the original branch from starting again? I've tried changing the order of rewritehostport and append_branch without any difference. I've also tried to use t_relay() instead of the RELAY route. I'm using Kamailio 5.1 and have modified the deafult kamailio.cfg Thanks! Boden I think this is the relevant parts of the config: # PSTN GW routing route[PSTN] { [PSTN regex logic omitted] # classify as an outbound call set_dlg_profile("outbound_calls"); # select a PSTN gateway from the dispatcher if (!ds_select_dst("66", "8")) { send_reply("404", "No destination"); exit; } # failover if the selected PSTN gateway is down t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destinations in failure route failure_route[RTF_DISPATCH] { xlog("inside RTF_DISPATCH"); if (t_is_canceled()) { exit; } # wrong number from PSTN if (t_check_status("404")) { append_branch(); append_hf("X-Application: announcement\r\n"); append_hf("X-Announcement-Playback: wrong-number\r\n"); route(TO_ANNOUNCEMENT); exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { xlog("selecting next destination"); if (ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Routing to announcement server route[TO_ANNOUNCEMENT] { if (!is_method("INVITE")) { return; } rewritehostport('ANNOUNCEMENT_SERVER:5060'); route(RELAY); exit; }

2 3

Problem with selection Transport Protocol - MultiHomed Problem
by YAS0 CANER 17 Dec '18

17 Dec '18

Hello , i have trouble about selection transport protocol. i have tried so many thing but couldnt find TM selected which IPv? and protocol type (TCP or UDP) For example; i have 2 interface ipv6 and ipv4. i will route my message (INVITE-Register etc.) another node that use TCP and IPV6 , domain name is test1.test. TM module or some module resolve this domain to IPV6:UDP but i want to sent it over IPV6:TCP socket. so i cannot know which protocol is resolved and cannot route the right socket. $du variable show only sip:ali@test1.test Conclusion : There isn't any pseudo value about which destination of IP version will use. Thanks Flow: UAC (IPv4:UDP) ----------> Kamailio (MultiHomed) ----> (TCP:IPv6)Kamailio ---- Kamailo (multiHomed ) cfg ------ auto_bind_ipv6=1 dns_try_ipv6=yes dns_retr_time=1 dns_retr_no=1 dns_cache_flags=4 mhomed=1 fork=yes tcp_source_ipv6 = [fd61:c23c:b3b6::5] listen=tcp:[fd61:c23c:b3b2::5]:5060 listen=udp:[fd61:c23c:b3b2::5]:5060 listen=udp:192.168.1.107:5060 listen=tcp:192.168.1.107:5060 route []{ if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 1\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } else if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 2\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else if($dP=="UDP"){ xlog("L_INFO","[route] [UDP] 1\n"); if (!t_relay_to_udp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else{ xlog("L_INFO","[route] [??] 1\n"); if (!t_relay()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } }

2 5

Re: [SR-Users] sr-users Digest, Vol 163, Issue 15
by Toufic Mobarak 17 Dec '18

17 Dec '18

Tnhxtx Regards, Toufic Mobarak > On Dec 15, 2018, at 6:00 AM, sr-users-request(a)lists.kamailio.org wrote: > > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Routing of Audio and Video Stream. (Daniel-Constantin Mierla) > 2. Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 3. Re: Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 4. Re: Routing of Audio and Video Stream. (Sergiu Pojoga) > 5. Re: Problem with selection Transport Protocol - MultiHomed > Problem (Daniel-Constantin Mierla) > 6. Issue on kamailio 4.1.9 of database reconnection (Emmanuel BUU) > 7. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > 8. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 14 Dec 2018 12:54:44 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, Abhijit Dutta <duttaa(a)hotmail.com> > Subject: Re: [SR-Users] Routing of Audio and Video Stream. > Message-ID: <26c51f12-5242-4794-5124-f37bc523c7ae(a)gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > > I guess you still need to send the INVITE to Asterisk, right? > > > Cheers, > Daniel > > >> On 14.12.18 11:15, Abhijit Dutta wrote: >> Hi All, >> >> Is it possible to redirect only the audio stream towards Asterisk from >> Kamailio (Proxy) via RTPEngine and not to send the video stream to >> Asterisk. I am yet to get the complete reason why my client wants this >> to happen though. >> >> Thanks >> Abhijit >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com > Kamailio Advanced Training -- www.asipto.com > >

2 1

Automating installation
by John Tuxies 14 Dec '18

14 Dec '18

Hi all. I am new to this and i would like some help please. I am looking to deploy Kamailio and Asterisk in my systems and i would like to automate the process. I have found a bash script but it it outdated from here: https://github.com/gurumelo/asteriskkamailio i did some changes and modifications to adapt it to newer version of software, but it fails to start Kamailio and continue to Asterisk. I would appreciate if someone could help me with this script and make it work. The systems i am planning to execute it is in Debian 9 and Ubuntu 18.04 server, both 64bit. Sincerely yous, John

8 21

FreeBSD Kamailio pkg
by Sergiu Pojoga 14 Dec '18

14 Dec '18

Did anyone notice that when installing Kamailio from FreeBSD package, by default, it accepts registration with whatever account credentials? Not sure if it's like that with other OS packages (usually I install it from source), but this behavior kind of doesn't go with the popular saying that Kamailio by default "does nothing", lol. Regards, --Sergiu

3 4

Kamailio ERROR: _reply_light(): can't generate xxx reply when a final yyy was sent out
by Olli Attila 13 Dec '18

13 Dec '18

Hello, I'm getting these error messages on kamailio log from time to time: Dec 11 19:18:30 /usr/sbin/kamailio[23913]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 200 was sent out Dec 8 04:44:56 /usr/sbin/kamailio[14960]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 486 was sent out Dec 5 07:46:04 /usr/sbin/kamailio[14961]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 480 was sent out Kamailio is configured to act as a proxy between SBC and a softswitch. According to the signaling we can see that these errors occure when the SBC sends CANCEL message towards Kamailio when at the same time final reply is received from the Softswitch side. The call seems to be teared down nicely and finally Kamailio replies to SBC by sending "200 ok, no more pending brances". I guess Kamailio treats this situation as an error because the sip processing engine has sent the final reply out and at the same time SBC sends the CANCEL to Kamailio. Is there any way we could handle this situation better on Kamailio end? The three example ERROR messages are from cases where B subscriber has answered the call, the B has been busy and the last one is from a case where B subscriber has been temporarily unavailable. Cheers, Olli

2 4

is_e164 logic to detect valid number
by Patrick Wakano 11 Dec '18

11 Dec '18

Hello list, Hope you are all doing fine! I am using the is_e164() function to validate the number we receive, and I come to see that the number +555 was accepted.... After some googling it looks like(it is not very clear though) that 7 digits are the minimum we could have for e164 numbers but after checking the source code, I saw it accepts anything starting with + and having between 2 and 16 numbers. So is it really valid to have a number with just 2 digits? What is the case? Thank you, Kind regards, Patrick Wakano

5 7

Performance tests of native vs. kemi scripting for SIP routing
by Daniel-Constantin Mierla 11 Dec '18

11 Dec '18

Hello, during the preparations of Kamailio v5.2.0 and the days after, I got the time to run some tests in order to see the differences of running similar SIP routing logic when using native configuration file scripting versus KEMI languages. So far I did it for Lua and Python (v2), hopefully I will find time to run also for the rest of supported scripting languages: Python (v3), JavaScript, Ruby, Squrrel and Lua with LuaJIT. The pleasant surprise was to see that this time, the results of using Python were similar to native scripting and Lua. When I did some basic tests during development of KEMI, using Python seemed slower. More details about how tests were performed, the config files/scripts, used tools and results were published in a wiki page: - https://www.kamailio.org/wiki/kemi/performance-tests/5.2.x A news article with further details is on kamailio website: - https://www.kamailio.org/w/2018/12/performance-tests-for-kemi-scripting-lan… Should anyhow of you have some time to run the tests in own environment and share the results (on sr-users mailing list), it would be appreciated and would help to validate the tests and discover if I forgot something in any of the configs. Note that the goal of the tests is not measuring the capacity of Kamailio in your environment, but see the differences between using different scripting languages in the same environment (so you can do the tests on raspberry pi, if you like). I would also be interested in knowing if you are using Kamailio with KEMI or you are planning to, if yes, which scripting language. If already doing it, is already in production? These are useful to plan where to allocate developer resources in the future... And to start, I am running Kamailio with Lua (4 instances) and Python (1) scripting in production deployments, with very good results so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training -- www.asipto.com

2 2

core at kamailio 5.1.6
by David Escartín 11 Dec '18

11 Dec '18

hello all we are seeing in a new kamailio instance some crashes. We are not sure the root cause, here you have the last core generated https://pastebin.com/xMRrug2X COuld you please take a look to see if there might be any issue with the last message that caused the core (to me seems normal) or there might be some issues related to other system libraries, etc? thanks a lot and regards david

3 7

DTLS HA redis-pacemaker-corosync
by Жан Базаров 10 Dec '18

10 Dec '18

Hello! Are there any ways to implement using DTLS in redis HA scenario? When I reboot the active RTPEngine server the system correctly fails over to the other server and, after only a small pause (~1s), incoming audio continues to arrive. However, approximately 15 seconds after this I hear nothing but silence. Thanks.

2 1

Kamailio World 2019 - Call for Presentations
by Daniel-Constantin Mierla 10 Dec '18

10 Dec '18

Hello, I would like to announce that Call for Presentations at Kamailio World 2019 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k07/call-for-speakers/ The 7th edition of the event takes place again in Berlin, Germany, during May 6-8, 2019. Expect over 150 participants, developers and community members as well as representatives from other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training -- www.asipto.com

1 0

SIP SIMPLE TO XMPP AND VICE VERSA
by sagar malam 08 Dec '18

08 Dec '18

Hello, We have a system which using FreeSWITCH for calls and Ejabberd( XMPP) for chat.Now we want to support SIP SIMPLE as well.But we are not going to move old users to SIP SIMPLE but only new users will use SIP SIMPLE.This leads to requirement of having SIP-XMPP gateway for which are are considering Kamailio server. I have gone through documentation about Kamailio XMPP module, it is clear that Kamailio can convert SIP MESSAGE to XMPP so SIP to XMPP will work but what about XMPP to SIP MESSAGE ? Does that work as well ? Thanks in advance. -- Thanks, Sagar

2 2

Dialplan Module limits
by Enrico Bandiera 07 Dec '18

07 Dec '18

Hi guys, does anyone know which are the limits of the diaplan module? After how many entries I will encounter serious performance issues? Thanks, Enrico.

2 1

question about replace_body_all
by Wilkins, Steve 07 Dec '18

07 Dec '18

Hello, I am attempting to change the Message Body of an SIP message using replace_body_all; I want to replace all attributes that start with rtcp. Here is my code => replace_body_all("rtcp:{1}[0-9]{5,}", newstring), and I call msg_apply_changes() after this call. Note that I verified that the regex is working using search. The problem is that nothing gets changed. However, what I do get is a new string at the bottom of the Message Body with my newstring concatenated three times, which happens to be the number of times that rtcp occurs in the Message Body. Note, I am doing this so that I can add the IP address to the rtcp attribute because this is required by a Provider. Any ideas? Thank you, -Steve

3 2

Dialog getting disconnected as soon as call is ended
by Subham Singhal 07 Dec '18

07 Dec '18

Hello, I am new to kamailio. I have been trying to integrate kamailio with asterisk. I am able to send INVITE and get RTP packets as well, but I am unable to receive BYE from asterisk. Call is getting terminated through following websocket error. "WebSocket connection to 'wss://172.16.16.169:5061/' failed: One or more reserved bits are on: reserved1 = 1, reserved2 = 0, reserved3 = 0." -- Thanks and Regards Subham Singhal

2 1

Auth_xkeys module - freeing already freed pointer
by José Seabra 06 Dec '18

06 Dec '18

Hello there, I'm constantly getting the following message when i'm using module auth_xkeys. CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7f6310987400), called from core: core/data_lump.c: free_lump(466), first free core: core/data_lump.c: free_lump(466) - ignoring Apparently kamailio only throw this message but i would like to know if its a normal protection from kamailio or it should be fixed? Thank you for the support Regards -- José Seabra

3 5

Assistance with re.subst & newline
by Mark Hall 06 Dec '18

06 Dec '18

Hi, I am using rtpengine to proxy audio to/from media servers. The outbound calls are being rejected by a couple of my carriers (although accepted by others), and the closest that I have been able to find for the reason is the inclusion of "a=rtcp:xxxxx" in the SDP generated by rtpengine. To test further, I am trying to remove this line. Rtpengine itself doesn't have any options to remove it, so I am left with manipulating the generated SDP in Kamailio. A sample of generated SDP is: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv a=rtcp:30939 s=My Media Server I am using rtpengine_manage("RTP/AVP ICE=remove"); to generate the SDP to a variable ($avp(sdp)) Now, I am trying to just remove the whole a=rtcp line with re.subst $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*\R||s}); *** Tried / as delimiter also, but | looks cleaner *** This should match the line and any CR/LF characters - BUT in results, nothing is changed (various online Regex testers show this match to be correct - I tried \r\n also). If I do the same but without the \R, then the line is removed but not the CR/LF, leaving a gap in the SDP - which is invalid. $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*||s}); Gives: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv s=My Media Server Can anyone see where I am going wrong, or suggest an alternative? Kind Regards, Mark Hall

3 3

New features: dialog - list dialogs via rpc using matching expressions
by Daniel-Constantin Mierla 06 Dec '18

06 Dec '18

Hello, being discussed and requested couple of times via lists, irc channel and discussions at events world wide, I just want to send a short note to announce that yesterday I pushed a new rpc command to dialog module that returns dialog records that match on conditions applied to request uri, from uri, to uri or callid fields. The filtering can be done with string comparison, regular expression or starts-with (prefix) matching. See more details at: - https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.list_… It should enable listing dialogs only for specific users, caller or callee, without dumping all dialogs. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 0

set_advertised_address
by Kjeld Flarup 06 Dec '18

06 Dec '18

Hello I have a PBX behind NAT. Thus I advertise the public IP, and forwards the port to my PBX. listen=LOCALIP:5070 advertise EXTERNALIP:5070 Now clients can connect to the PBX from the Internet. And also inside the LAN, because I have enabled NAT loopback. However some customers sysadmins complains that NAT loopback is a security risk. I have not been able to find any exploits of this, but the sales and support people asks if it is possible to remove this NAT loopback requirement. I could look at $rd and if it is local, then I could advertise LOCALIP. I found set_advertised_address("LOCALIP"); set_advertised_address however only seems to modify the latest Via header, not the Record-route, and audio neither works. Could I do something to make this work, or is it a dead end? -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 11

Dispatcher clear dst_avp
by Grant Bagdasarian 05 Dec '18

05 Dec '18

Hello, I'm trying to use multiple destination sets which may include the same destination. For instance: SET ID: 1 sip:10.0.0.1:5060 sip:10.0.0.2:5060 SET ID: 2 sip:10.0.0.1:5060 Whenever a new SIP INVITE is sent to the Kamailio instance it will use SET ID 1. If then for some reason this call attempt fails, failure_route will trigger and SET ID 2 will be used. At first glance this might not make any sense, but there is some business logic behind this, but it's not relevant for this case. For some reason, the dst_avp variable still contains the destinations of SET ID 1 in failure_route. Is there a built-in way to reset this variable for every call to ds_select_domain(), such that only the destinations of passed SET ID will be loaded into the avp? I tried setting the avp to $null before calling ds_select_domain, but that had some undesirable effects. The reason I'm asking this is the destinations the call is being sent to are Freeswitch instances, which will return a 482 Request Merged when the same INVITE is sent to the same Freeswitch within 4 seconds (default T4 timer). I'm trying to match the previously selected destination in failure_route with the next destination and skip if they're the same, or stop processing when they are the same and no more destinations are available. I know I can mess with the T4 timer in Freeswitch and perhaps even sleep in failure route to deal with the 482 failure, but I rather fix this in a normal way. Regards, Grant Bagdasarian Senior Developer

2 2

xlog issue
by Soltanici Ilie 04 Dec '18

04 Dec '18

Hello, I have a strange situation by using xlog module. I don't know for what reason I'm not receiving all logs generated by kamailio. This is the configuration which I'm using in kamailio.cfg: log_facility=LOG_LOCAL0 log_name="kamailio" log_prefix="{$rm ($mt) | Seq=$cs | Source IP=$si ($proto) | Call ID=$ci} " # ----- xlog ----- modparam("xlog", "buf_size", 8192) modparam("xlog", "long_format", 0) modparam("xlog", "force_color", 1) modparam("xlog", "log_colors", "L_ERR=cr;L_WARN=px,L_INFO=px") This is rsyslog configuration file: local0.* /var/log/kamailio/kamailio.log Kamailio is running under kamailio user, permission for the log file are as shown below: -rwxrwxr-x 1 kamailio kamailio 252885132 Dec 2 14:51 /var/log/kamailio/kamailio.log In request_route block this is the first line: xlog("L_INFO","[START ROUTING]-(Source IP=$si:$sp/Destination IP=$Ri:$Rp)\n"); The problem is that not every request is logged in the log-file. For example, some "INVITES" requests I can find in the log file, but some of them - I cannot, even that in sngrep I see the request and the call is successfully processed by kamailio. Also, for some requests i can see only partial data, not full call-flow as it supposed to be (for ex. i see only BYE requests, or ACK response instead of full call flow). Does someone have the same issue? If you don't - how are you dealing with kamailio log files? I'm thinking to send them to the central ELK stack, but if I have such problems by saving them locally - I don't see any reason to send them elsewhere. I may think that the problem could be in rsyslog itself, but how can I troubleshoot that? The traffic on the server is not very high - 30-50 concurrent calls. As a storage i'm using an SSD disk and xfs filesystem. Load on the disk - according to iostat/iotop - is minimum. OS: CentOS Linux 7 (Core), Kamailio: 5.2.0 (x86_64/linux) 535e13 Thank You.

3 9

presence/mwi issue
by Andreas Granig 04 Dec '18

04 Dec '18

Hi, Currently playing with presence_mwi in kamailio 5.1, and it appears the issue reported by Juha two years ago at https://lists.kamailio.org//pipermail/sr-dev/2016-September/036609.html seems to have appeared again somewhere along the way. I’m sending a PUBLISH with message-summary event type, however I’m getting this error: ERROR: presence [presentity.c:283]: check_if_dialog(): failed to parse xml document As said in the old thread, it’s not an XML document, so it shouldn’t try to treat it as such. Any ideas? Best, Andreas

2 8

Dispatcher with a naptr address as destination
by José Seabra 04 Dec '18

04 Dec '18

Hello all, I'm trying to use a naptr address as a dispatcher destination and i'm facing some issues here. When i configure a naptr address in dispatcher table, like: *1:1:sip\:proxy-registrar-ci1.fuze.domain:0:0:'':''* I see kamailio making a dns query of type A, but since that this name doesn't have any A record configured in dns, kamailio didn't get any positive reply from it. Should't kamailio make dns query by this order? 1. NAPTR 2. SRV 3. A I noticed also that if the name configured in dispatcher table is resolved successfully with A record from dns, then kamailio tries to make a query to NAPTR and then SRV services for the exactly same name used for query previously done(type A), but only if it gets a positive answer from dns to the A type query. I'm using kamailio version 5.1.0 with the following dns parameters configured: ####### DNS Parameters ######### dns = no rev_dns = no dns_try_ipv6 = no dns_retr_time = 1 dns_retr_no = 1 dns_cache_del_nonexp = no dns_cache_flags = 1 dns_cache_gc_interval = 120 dns_cache_init = 1 dns_cache_mem = 1000 dns_cache_negative_ttl = 60 dns_try_naptr = yes use_dns_cache = on use_dns_failover = on dns_srv_lb = yes Is there something wrongly configured here that can make kamailio behaves like that? Thank you. Regards -- Cumprimentos José Seabra

4 8

Issue with ACK's
by Dan Quinney 30 Nov '18

30 Nov '18

Hi all, I’m having some issues with ACK’s. Specifically, when an ACK reaches Kamailio from either Asterisk or an endpoint, Kamailio is adding a second Via header - which isn’t valid as per RFC 3261 (17.1.1.3); “The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request." Endpoints are registered with Kamailio but INVITES are routed via Asterisk for dial plan and media reasons. I’m running fairly bog-standard Kamailio and Asterisk installs. I’ve attached a sample PCAP for if anyone has any suggestions. Thanks. Dan

2 2

Issue with ACK's
by Dan Quinney 30 Nov '18

30 Nov '18

Hi all, I’m having some issues with ACK’s. Specifically, when an ACK reaches Kamailio from either Asterisk or an endpoint, Kamailio is adding a second Via header - which isn’t valid as per RFC 3261 (17.1.1.3); “The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request." Endpoints are registered with Kamailio but INVITES are routed via Asterisk for dial plan and media reasons. I’m running fairly bog-standard Kamailio and Asterisk installs. I’ve attached a sample PCAP for if anyone has any suggestions. Thanks. Dan

1 0

Kamailio v5.2.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Nov '18

30 Nov '18

Kamailio v5.2.0 is out – it comes with 6 new modules and a consistent group of improvements touching more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.2.0 brings more flexibility and optimizations for KEMI interpreters (including two new interpreters for Python3 and Ruby), major enhancements to load balancer and tls implementations, new variables and lots of other new features. Enjoy Kamailio v5.2.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2019 - http://www.kamailioworld.com

2 2

Response VIA Header
by Soltanici Ilie 29 Nov '18

29 Nov '18

Hi, We have a Kamailio Instance running on Public IP Address, one of our ISP cannot send ACK back to us because we are sending 100 Trying - without "received" parameter. Is there any way in Kamailio to force a "received" parameter in the via header for 100 Trying Message? SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK8tgchsfjaKJdf.1;rport=5060 Thank you.

2 1

Forward() not handles Route headers
by Yuriy Gorlichenko 29 Nov '18

29 Nov '18

We are building some stateless solution and found that in case of using forward() kamailio sends traffic only via URI or $du variable. It never removes itself from route headers. Is this is the truth? Or I need to add some other module to handle it? (rr already added)

1 1

KAMAILIO 4.2 - RTPPROXY config confirmation - error message from systemctl status kamailio.service -l (ERROR: <script>: NATMANAGE coee)
by yboujraf＠by-research.be 28 Nov '18

28 Nov '18

Dear All, I am still using Kamailio 4.2 with Debian Jessie without any issues. RTPPROXY - CONFIG > # Defaults for rtpproxy > # The control socket. > #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > # To listen on an UDP socket, uncomment this line: > CONTROL_SOCK=udp:127.0.0.1:7722 > # Additional options that are passed to the daemon. > > # log in /var/log/kamailio.log > EXTRA_OPTS="-l ip_public/192.168.40.101 -m 11000 -M 12000 -d DBUG:LOG_LOCAL0" WHEN I MAKE AN INVITE FROM 801 TO 803, I HAVE THIS ERROR MESSAGE : KAMAILIO SERVICE (ERROR: <SCRIPT>: NATMANAGE COEE) root@sipsecure:~# systemctl status kamailio.service -l ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio) Active: active (running) since Tue 2018-11-20 22:34:31 CET; 40s ago Process: 1820 ExecStop=/etc/init.d/kamailio stop (code=exited, status=0/SUCCESS) Process: 1830 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) CGroup: /system.slice/kamailio.service ├─1846 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1848 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1849 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1850 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1851 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1852 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1853 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1854 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1855 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1856 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1858 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1860 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1863 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1865 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1867 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1869 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1871 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1873 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio └─1875 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1854]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1854]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1855]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1855]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: transaction answered: timestamp=1542749703;method=INVITE;from_tag=4d488551;to_tag=2006d156;call_id=fjpUwRGm0u--3RvvB4fZ7Q..;code=200;reason=OK;src_user=801;src_domain=192.168.40.101;src_ip=192.168.0.229;dst_ouser=803;dst_user=803;dst_domain=192.168.0.207;user_agent=Z 5.2.19 rv2.8.99 Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE M=ACK OU=sip:803@192.168.0.207:5060;alias=192.168.0.207:5060~1 RURI=sip:803@192.168.0.207:5060 RD=192.168.0.207 F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=192.168.0.207 IP=192.168.0.229 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE coee Don't understand why the error??? RTPPROXY SERVICE : (RTPPROXY IS DOING HIS JOB PERFECTLY AND GET AUDIO ON BOTH DIRECTION) root@sipsecure:~# systemctl status rtpproxy.service -l ● rtpproxy.service - LSB: RTP Proxy Loaded: loaded (/etc/init.d/rtpproxy) Active: active (running) since Tue 2018-11-20 22:13:58 CET; 22min ago Process: 443 ExecStart=/etc/init.d/rtpproxy start (code=exited, status=0/SUCCESS) CGroup: /system.slice/rtpproxy.service └─1137 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 212.76.231.102 192.168.40.101 -m 11000 -M 12000 -d DBUG LOG_LOCAL0 Nov 20 22:34:31 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1875_11 1 " Nov 20 22:35:00 sipsecure rtpproxy[1137]: DBUG:handle_command: received command "1853_8 USEEc106,9,3,111,0,8,97,110,112,98,101,100,99,102 fjpUwRGm0u--3RvvB4fZ7Q.. 192.168.0.229 11000 4d488551;1" Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: new session fjpUwRGm0u--3RvvB4fZ7Q.., tag 4d488551;1 requested, type strong Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: new session on a port 11496 created, tag 4d488551;1 Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: pre-filling caller's address with 192.168.0.229:11000 Nov 20 22:35:00 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1853_8 11496 192.168.40.101 " Nov 20 22:35:03 sipsecure rtpproxy[1137]: DBUG:handle_command: received command "1852_8 LSEEc106,9,3,111,0,8,97,110,112,102,98,101,100,99 fjpUwRGm0u--3RvvB4fZ7Q.. 192.168.0.207 11112 4d488551;1 2006d156;1" Nov 20 22:35:03 sipsecure rtpproxy[1137]: INFO:handle_command: lookup on ports 11496/11402, session timer restarted Nov 20 22:35:03 sipsecure rtpproxy[1137]: INFO:handle_command: pre-filling callee's address with 192.168.0.207:11112 Nov 20 22:35:03 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1852_8 11402 192.168.40.101 " ??? Question : for rtpproxy config, do I need to add the natted ip address (NAT 1:1) or the real public address (where ip phone needs to assign as domain)? I am confused. Thanks a lot in advance Best Regards, Youssef

3 6

Issue sending SIP OPTIONS request to multiple contacts
by Dan Quinney 28 Nov '18

28 Nov '18

Hi, When using Kamailio v5.2.0-rc2 (x86_64/linux) 9e729b and setting the below nathelper module parameters it appears only the first contact is sent a SIP OPTIONS request. modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@mydomain <sip:ping@mydomain>.com") modparam("nathelper", "keepalive_timeout", 90) Reading the release notes for v5.2 what I’m doing *should* be possible, but as the functionality has been recently added I’m assuming that it might be a bug. Any assistance on this would be great. Thanks, Dan

2 1

Releasing Kamailio v5.2.0
by Daniel-Constantin Mierla 28 Nov '18

28 Nov '18

Hello, I am consider to release v5.2.0 (the first stable version out of branch 5.2) next week, likely on Wednesday, Nov 28, 2018. It still allows a bit more than a week of testing as well as well time to prepare the online resources for it (documentation, wiki pages, upgrade guidelines, etc...). If there is any issue you are of and not yet reported to github bug tracker, do it as soon as possible to give it a chance to be fixed in time for the next major release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 2

Error opening Kamailio's FIFO
by Markus 28 Nov '18

28 Nov '18

Hi list, when I run "/usr/sbin/kamctl address reload" I get: ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. In /etc/kamailio/kamailio.cfg I have: loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") But there's no fifo in /tmp. What am I doing wrong? kamailio-4.4.4-1.1.x86_64 on CentOS Linux release 7.2.1511 (Core). Thank you! Markus

3 8

[CFP] FOSDEM 2019, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 28 Nov '18

28 Nov '18

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2019 takes place 2-3 February 2019 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 3rd of February 2019. The lounge will be present for both days. To discuss the dev-room and lounge, please join the Free RTC mailing list: http://lists.freertc.org/mailman/listinfo/discuss To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 2nd of December. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM19/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2019/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 2nd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 1st of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: http://lists.freertc.org/mailman/listinfo/discuss Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: http://lists.freertc.org/mailman/listinfo/discuss The dev-room administration team: Saúl Ibarra Corretgé <s(a)saghul.net> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro> Guus der Kinderen <guus.der.kinderen(a)gmail.com>

2 1

Dialiog and dispatcher
by David Villasmil 27 Nov '18

27 Nov '18

Hello guys, I'm testing the dispatcher with the dialog module. After making several test calls, which are properly routed, the number of dialogs stay there for a while, preventing new calls from going to those destinations: "SET": { "ID": 101, "TARGETS": [{ "DEST": { "URI": "sip:1.2.3.4:5080", "FLAGS": "AP", "PRIORITY": 0, "ATTRS": { "BODY": "weight=100;maxload=10;duid=bedd39c7-6c73-4477-ae7d-77f8cdc96797", "DUID": "bedd39c7-6c73-4477-ae7d-77f8cdc96797", "MAXLOAD": 10, "WEIGHT": 100, "RWEIGHT": 0, "SOCKET": "" }, "RUNTIME": { "DLGLOAD": 10 } } }] The calls have been down for several seconds, but the "DLGLOAD" parameter doesn't go down, even though /usr/local/kamailio5/sbin/kamctl', 'dialog', 'show Output the correct amount of dialogs. I set the dialog handler just before relaying out: t_on_failure("RTF_DISPATCH"); set_dlg_profile("gateways","$du"); route(RELAY); And the module is supposed to remove them automatically. Am i doing something wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

dispatcher: Get gateway current calls
by David Villasmil 27 Nov '18

27 Nov '18

Hello guys, is there any way of printing out the selected gw current calls? i.e.: if(ds_select_dst("1", "4")) { xlog("L_ERR", "[DISPATCH] $ci: ds_select_routes was succesful (du: <$du>)\n"); ****** PRINT GWs CALLS ******* } else { xlog("L_ERR", "[DISPATCH] $ci: did NOT find a gateway to use!\n"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "[DISPATCH] $ci: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Re: [SR-Users] "kamdbctl create" failing with MySQL 8
by Jonathan Tyler 26 Nov '18

26 Nov '18

I just had the same thing happen to me. I installed Kamailio via Apt, and I'm using a dedicated MySQL server, with non-root admin credentials. Previously today I installed it using a local MySQL 5.7 database with no issues. Kamailio Host: Kamailio 5.1.2 (Ubuntu 18.04) MySQL Host: MySQL 8.0.13 (Windows Server 2012 R2) It creates the database, then fails at granting permissions. Accounts were created and privileges flushed before executing. > root@kamailio:~# kamdbctl create > INFO: creating database kamailio ... > mysql: [Warning] Using a password on the command line interface can be insecure. > INFO: granting privileges to database kamailio ... > mysql: [Warning] Using a password on the command line interface can be insecure. > ERROR 1064 (42000) at line 1: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'IDENTIFIED BY '<password>'' at line 1 I turned on query logging on the database host but didn't see any errors there, or in the error log itself. > 2018-11-21T01:58:15.045586Z 8 Connect <user>@kamailio on using SSL/TLS > 2018-11-21T01:58:15.046239Z 8 Query select @@version_comment limit 1 > 2018-11-21T01:58:15.046764Z 8 Query CREATE DATABASE kamailio CHARACTER SET utf8mb4 > 2018-11-21T01:58:15.189109Z 8 Quit > 2018-11-21T01:58:15.208476Z 9 Connect <user>@kamailio on using SSL/TLS > 2018-11-21T01:58:15.209019Z 9 Query select @@version_comment limit 1 > 2018-11-21T01:58:15.210114Z 9 Quit I've tried with the default MySQL 5.7 Client on Ubuntu, and upgrading it to 8.0.13, but got the same result. Unfortunately I don't know how to see the output of the SQL Statement being sent (without installing 5.7), or I'd try to run it manually. - Jonathan

3 5

issue tracking dialogs with db_redis in kamailio 5.2
by David Escartín 26 Nov '18

26 Nov '18

hello all using version 5.2.0-rc0 of kamailio, we are trying to track the dialogs of dialog module using db_redis module with this setup #!define DBURL_DIALOG "redis://127.0.0.1:6390/7" # ----- dialog params ----- modparam("dialog", "db_url", DBURL_DIALOG) modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 16384) modparam("dialog", "dlg_flag", 31) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "timer_procs", 1) modparam("dialog", "db_update_period", 1) # ----- db_redis params ----- modparam("db_redis", "schema_path", "/usr/local/kamailio/share/kamailio/db_redis/kamailio") modparam("db_redis", "keys", "dialog=entry:hash_entry,hash_id,callid") modparam("db_redis", "keys", "dialog_vars=entry:hash_entry,hash_id,dialog_key,dialog_value") and we see it works and if a kamailio instance resets in the middle of a dialog it recovers all the dlg_vars and values because we have the db_skip_load to 0 in dialog module. But we have noticed that the redis is being written with the information of the dlg_var assigned until that moment when we get a provisional response with SDP, and we dom see a set on the redis with the dialog entry and id untill call is connected. Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: INFO: Call-ID 1-1511(a)9.7.8.1: Status 180 Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' on the redis (we only see hmset for dialog_vars key) 1542374575.721816 [7 127.0.0.1:56858] "HMSET" "dialog_vars:entry::13725:16494:_uac_funew:sip:1231215423@voda.interconnect.manxtelecom.im" "hash_entry" "13725" "hash_id" "16494" "dialog_key" "_uac_funew" "dialog_value" "sip:1231215423@voda.interconnect.manxtelecom.im" 1542374575.722001 [7 127.0.0.1:56858] "HMSET" "dialog_vars:entry::13725:16494:_uac_fu:sip:anonymous@voda.interconnect.manxtelecom.im" "hash_entry" "13725" "hash_id" "16494" "dialog_key" "_uac_fu" "dialog_value" "sip:anonymous@voda.interconnect.manxtelecom.im" .... Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: INFO: Call-ID 1-1511(a)9.7.8.1: Status 200 Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog_vars' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 2 is 'dialog_key' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: INFO: Call-ID 1-1511(a)9.7.8.1: ACK received in A-Leg on the redis 1542374585.810520 [7 127.0.0.1:56882] "SCAN" "0" "MATCH" "dialog:entry::*" "COUNT" "1000" 1542374585.810672 [7 127.0.0.1:56882] "EXISTS" "dialog:entry::13725:16494:1-1511@9.7.8.1" 1542374585.810683 [7 127.0.0.1:56882] "HMGET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "hash_entry" "hash_id" 1542374585.810776 [7 127.0.0.1:56882] "HMSET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "state" "4" "timeout" "1542377585" "caller_cseq" "1" "callee_cseq" "0" "caller_contact" "sip:0000123456@9.7.8.1:5084" "callee_contact" "sip:50622959898@9.70.1.52:5060;transport=udp" 1542374585.810899 [7 127.0.0.1:56882] "SCAN" "0" "MATCH" "dialog:entry::*" "COUNT" "1000" 1542374585.810969 [7 127.0.0.1:56882] "EXISTS" "dialog:entry::13725:16494:1-1511@9.7.8.1" 1542374585.810977 [7 127.0.0.1:56882] "HMGET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "hash_entry" "hash_id" 1542374585.811059 [7 127.0.0.1:56882] "HMSET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "state" "4" "timeout" "1542377585" "caller_cseq" "1" "callee_cseq" "0" "caller_contact" "sip:0000123456@9.7.8.1:5084" "callee_contact" "sip:50622959898@9.70.1.52:5060;transport=udp" i don't see a redis set order to update the dialog entry key before the 200OK is received, to create and update state 1, etc. when created with dlg_manage command could you please check if we might be missing something? thanks a lot and regards david

2 3

Kamailio "Presence Server" configuration
by denislav.tsonev＠musala.com 26 Nov '18

26 Nov '18

Hello, I'm trying to configure Kamailio to be "Presence Server". Simple scheme: two phones (one "hard phone" Grandstream GXP1620 and "Soft phone" MicroSIP) and Kamailio server with default configuration. So when "MicroSIP" send SIP SUBSCRIBE to Kamailio, server response with SIP NOTIFY but without XML part. Then when hard phone change his status from "Ready" to "Ringing" and then "On-The-Phone" Kamailio is not sending any SIP NOTIFY messages. Is this normal behavior or I missed something? (In DB I have records in "active_watchers" and "watchers" tables but not in "presentity" table) ====================================================================== ====================== Module configuration: ============================= ====================================================================== #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:10.0.6.123:5060") # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ====================================================================== ============================================================== ======================== Route: ================================ ============================================================== route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } ================================================================ Thanks in advance Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

2 1

Dispatcher on 5.2
by Igor Olhovskiy 26 Nov '18

26 Nov '18

Hi! Am I missing something or it’s a bug? Idea is following. I’m using dispatcher to load-balance calls. Using same weight destinations, order like in a file. So, dispatcher.list looks like 1 sip:172.28.11.6:5060 1 sip:172.28.11.4:5060 I’m using round-robin algo with ds_select_dst(«1», «4») But when it selects 172.28.11.4 and it for a reason not working, it will never come back to 172.28.11.6. Log: … Relaying packet du: sip:172.28.11.4:5060 … BYE/INVITE timeout with no reply … 49(59) WARNING: dispatcher [dispatch.c:2314]: ds_update_dst(): no xavp uri field in next destination record ((nil)) modparam("dispatcher", "list_file", "/tmp/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_hash_size", 9) Kamailio: version: kamailio 5.2.0-pre0 Regards, Igor

2 3

database connections per child process
by Karthik Srinivasan 23 Nov '18

23 Nov '18

Hello, Is there a formula/algorithm that determines how many TCP connections are made per child process to a database server defined by sqlops' sqlcon parameter? ex: if i have 1 database on a database server defined by sqlcon, and i have 1 UDP child process, i notice that the child process has made 2 TCP connections to the database server. ex: if i have 3 databases on the same database server defined by 3 sqlcon statements (1 pertaining to each database) and i have 4 UDP child processes, i notice that each child process makes 4 TCP connections to the database server. ex: if i have 3 databases on the same database server defined by 3 sqlcon statements (1 pertaining to each database) and i have 6 TCP child processes, i notice that each child process makes 4 TCP connections to the database server. Basically wanting to know what governs the number of tcp connections a child process makes to a particular database server. Thank you, Karthik

3 3

IMS: SMS over IP - how to get the originator's tel address?
by Ron McLeod 22 Nov '18

22 Nov '18

I am working on a VoLTE/IMS scenario to transfer text messages as SMS over IP (+g.3gpp.smsip feature) between UE's. This is a high-level view of the message flows between the endpoints (not showing CSCFs): UE1 IP-SM-GW |---- MESSAGE (SUBMIT) --------------->| |<------------------- 202 ACCEPTED ----| | | |<-------- MESSAGE (SUBMIT_REPORT) ----| |---- 200 OK ------------------------->| | UE2 | | | |<-------------- MESSAGE (DELIVER) ----| |---- 200 OK ------------------------->| The messages does get transferred from UE1 to UE2 successfully. Since this is GSM-style SMS, the OA (originating address) and DA (destination address) are specified using telephone numbers, not sip addresses. This is not a problem for the DA, since it is specified in the SUBMIT TPDU from the originating handset, but when the IP-SM-GW sends the DELIVER TPDU it needs to specify the OA so that the recipient knows who sent the SMS, and this was not provided in the MESSAGE with the SMS-SUBMIT. According to 3GPP TS 24.341 (http://www.qtc.jp/3GPP/Specs/24341-b20.pdf): "The IP-SM-GW will have to use an address of the SM-over-IP sender that the SC can process (i.e. an E.164 number). This address will come from a tel URI in a P-Asserted-Identity header (as defined in RFC 3325 [13]) placed in the SIP MESSAGE request by the P-CSCF or S-CSCF." Looking at the MESSAGE request sent from the S-CSCF to the IP-SM-GW, it looks like neither the P-CSCF nor the S-CSCF is providing the tel address in a P-Asserted-Identity header: MESSAGE tel:+3108012345680 SIP/2.0 Route: <sip:defaultapp@tas.core.ims1.test;lr>, <sip:iscmark@scscf.core.ims1.test:6060;lr;s=1;h=0;d=0;a=7369703a303031303330 31323334353637383940696d732e6d6e633030332e6d63633030312e336770706e6574776f72 6b2e6f7267> P-Served-User: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org>;sescase=orig;regstat e=reg f: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org>;tag=2703358633 t: <tel:+3108012345680> CSeq: 555874977 MESSAGE i: 2703358625_2324045032@2001:470:eb88:b021:27f:9507:589d:c201 Via: SIP/2.0/UDP [2001:470:EB88:150:0:0:0:22]:6060;branch=z9hG4bKb857.bd3345ba81ec10b1d11639d 9c7414606.0;i=a Via: SIP/2.0/TCP [2001:470:EB88:150:5054:FF:FE44:BFC9];branch=z9hG4bKb857.be53cc1c583b9c5f9dc ebf3306f2b7c1.0 v: SIP/2.0/UDP [2001:470:eb88:b021:27f:9507:589d:c201]:8906;rport=8013;branch=z9hG4bK355262 2488 Max-Forwards: 68 P-Access-Network-Info: 3GPP-E-UTRAN-FDD; utran-cell-id-3gpp=0010300030000001 c: application/vnd.3gpp.sms Allow: MESSAGE d: no-fork User-Agent: Xiaomi_Redmi 3S_Android6.0.1_9 l: 27 P-Charging-Vector: icid-value=49565300000000511100004B01000000; icid-generated-at=2001:470:EB88:150:5054:FF:FE44:BFC9 P-Asserted-Identity: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org> P-Visited-Network-ID: core.ims1.test Message Body GSM A-I/F RP - RP-DATA (MS to Network) Message Type RP-DATA (MS to Network) RP-Message Reference RP-Message Reference: 0x05 (5) RP-Originator Address RP-Destination Address - (3108012345680) RP-User Data Length: 14 TPDU (not displayed) GSM SMS TPDU (GSM 03.40) SMS-SUBMIT 0... .... = TP-RP: TP Reply Path parameter is not set in this SMS SUBMIT/DELIVER .0.. .... = TP-UDHI: The TP UD field contains only the short message ..1. .... = TP-SRR: A status report is requested ...0 0... = TP-VPF: TP-VP field not present (0) .... .0.. = TP-RD: Instruct SC to accept duplicates .... ..01 = TP-MTI: SMS-SUBMIT (1) TP-MR: 92 TP-Destination-Address - (8000) TP-PID: 0 TP-DCS: 0 TP-User-Data-Length: (5) depends on Data-Coding-Scheme TP-User-Data SMS text: HELLO Is this something that in configurable at the P-CSCF or the S-CSCF, or would it require now development? Thanks, Ron

2 3

issues with the debian repo for kamailio 5.2
by Aleksandar Sosic 22 Nov '18

22 Nov '18

Hi guys, I was trying to install the last 5.2 kamailio version editing the `/etc/apt/sources.list`: ``` deb http://deb.kamailio.org/kamailio52-nightly xenial main deb-src http://deb.kamailio.org/kamailio52-nightly xenial main ``` But as you can see there's only the 5.3 and 4.3 available: ``` root@e7845b3041eb:/# apt-cache madison kamailio kamailio | 5.3.0~dev0+0~20181114003826.5+xenial | http://deb.kamailio.org/kamailio52-nightly xenial/main amd64 Packages kamailio | 4.3.4-1.1ubuntu2 | http://archive.ubuntu.com/ubuntu xenial/universe amd64 Packages kamailio | 4.3.4-1.1ubuntu2 | http://archive.ubuntu.com/ubuntu xenial/universe Sources kamailio | 5.3.0~dev0+0~20181114003826.5+xenial | http://deb.kamailio.org/kamailio52-nightly xenial/main Sources ``` Any hints on how to solve this are very welcome. :) Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

5 6

t_set_retr and NOTIFY send after handle_publish
by Vasiliy Ganchev 21 Nov '18

21 Nov '18

Hi, list! I have a tricky problem, need advise from the community. The problem I need to solve: - have a SIP device, with a big number of BLFs configured, in some moment of time the device is slow to handle incoming SIP messages (a lot of NOTIFYs received), kamailio start retransmitting NOTIFYs, and the device get stuck. My idea is to change retransmits timeout for NOTIFYs that are sent out after handle_publish, so if get stuck, not to make "DoS" I have. What I tried: - t_set_retr(5000, 10000); before handle_publish => does not work - my tm is compiled with DTM_DIFF_RT_TIMEOUT, and t_set_retr works for e.g. INVITEs I have read the mailing list and also studied sources of presence and tm modules. I see that at the moment there is no way to solve what I want from config. My next steps are going to make some hacks in C code... at least to understand the code better and find some new insights Any other ideas? Comments/ideas are highly welcomed! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 3

Closing all ws connections
by Andrey Deykunov 21 Nov '18

21 Nov '18

Hi guys, I've implemented an rpc call that closes all WS connections: void ws_rpc_close_all(rpc_t *rpc, void *ctx) { ws_connection_t **list = NULL, **list_head = NULL; ws_connection_t *wsc = NULL; int ret; list_head = wsconn_get_list(); if(!list_head) return; list = list_head; wsc = *list_head; while(wsc) { LM_WARN("Closing connection\n"); ret = close_connection(&wsc, LOCAL_CLOSE, 1000, str_status_normal_closure); wsc = *(++list); } wsconn_put_list(list_head); } Please take a look. Can I call close_connection() and perform list iterations in this function not using WSCONN_LOCK? Thanks, Andrey

3 2

from Asterisk to Kamailio
by Valter Nogueira 21 Nov '18

21 Nov '18

Today, I use Asterisk as a SIP/RTP PROXY I proxy from customers Asterisks to a VOIP provider, in a multi-homed server. Now, I want to move to Kamailio without any rupture in customer's configuration. As anyone can imagine I am kind of lost. USER ACCOUNTS In Asterisk, I create a dynamic host account named ACCOUNT1 and I receive in *FROM HEADER sip:ACCOUNT1@customer_ip_address* In Kamailio, I have to define the account's domain like *kamctl add ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com>* SIP/RTP PROXY In Asterisk, I just dialout to the VOIP PROVIDER like *dial (SIP/VOIP_ACCOUNT/${EXTENSION})* Asterisk does all the magic (it is a B2BUA). It bridges the new call and media to the original call. Moreover, user don't know anything about how call are completed, nor how credentials are setup and soon. In Kamailio, I guess that I have to use nat, tm and rtpproxy modules and maybe uac. I am not sure how to setup it. Can someone send me a clue? Thank you, Valter

4 7

Kamailio-5.1.6 Crash
by Hamid Hashmi 20 Nov '18

20 Nov '18

I have been using the following version of kamailio Version: kamailio 5.1.6 (x86_64/linux) Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: unknown Compiled with: gcc 4.8.5 Compiled on: 11:48:33 Oct 11 2018 and I noticed today that the Kamailio service is crashing. [root@kamailiostandby /]# gdb core.17805 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-100.el7_4.1 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... [New LWP 17805] Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/SIP'. Program terminated with signal 11, Segmentation fault. #0 0x00007f636559e54b in ?? () "/core.17805" is a core file. Please specify an executable to debug. (gdb) list No symbol table is loaded. Use the "file" command. (gdb) bt #0 0x00007f636559e54b in ?? () #1 0x00007f63a9571810 in ?? () #2 0x0000000000000000 in ?? () Regards Hamid R. Hashmi

2 3

registrar::save() does not save 'received' field of the 'location' table
by Andrey Deykunov 20 Nov '18

20 Nov '18

Hi, We deploy kamailio in the cluster containing two nodes (Active and Passive). When Active node becomes Passive, our websocket clients continue sending REGISTER to this node. Passive node routes these requests to the Active node through the UDP, that causes the following warning message: 2018-11-14T04:43:31.070005-08:00 qapbx-ca1-1 kamailio[25597]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:64.78.52.88:5060, to ws:10.9.3.116:20033) Every registered client should be stored in the 'location': $var(save_val) = save("location", 0, "sip:$au@$td"); if ($var(save_val) <= 0) { xlog("L_ERR", "[$fU,R,$ci]: Registration saving failed, contact $ct"); sl_reply_error(); } else if ($var(save_val) == 3) { xlog("L_INFO", "[$fU,R,$ci]: Unregistration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } else { xlog("L_INFO", "[$fU,R,$ci]: Registration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } Log output of the Avtive node: 2018-11-14T04:43:31.063376-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=1, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.063947-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: Our node is active 2018-11-14T04:43:31.064323-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Authorization headers are empty in REGISTER command. Send 401 Unauthorized to 64.78.52.88 2018-11-14T04:43:31.285967-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=2, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.286564-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: Our node is active 2018-11-14T04:43:31.286943-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Registration accepted. Login: 54004221; ua: JsSIP 3.2.11; s-ip: 64.78.52.88; s-port: 5060; 3d parm: sip:54004221@qapbx-ca1.qaserverdata.net but in this scenario the save() method stores null value in the 'received' row. How can I fix it? Thank you, Andrey

2 1

RTPengine: shouldn't SDP <sess-version> be incremented when SDP is updated in-dialog?
by George Diamantopoulos 20 Nov '18

20 Nov '18

Hello all, I have the following issue (I think) with RTP engine. An INVITE comes in, and rtpengine will rewrite the SDP accordingly, as configured in kamailio.cfg. After some time a reINVITE is sent out in the opposite direction, for session refresh purposes. As I use rtcp-mux-offer in kamailio.cfg for this direction, RTPengine will inject the rtcp-mux parameter, and this reINVITE is forwarded to the UAC that sent the original INVITE. However, the SDP in the reINVITE is exactly the same as the SDP in the 200 OK to the original INVITE, with the exception of the rtcp-mux parameter. Since the SDP offered from the same end has changed, shouldn't session version be incremented as well? Does this sound like something that should be reported as a bug to rtpengine? Or am I missing something here? Thanks! Best regards, George Diamantopoulos

3 4

Number of sql connections
by Duarte Rocha 19 Nov '18

19 Nov '18

Greetings, I have my core parameter listen like this : listen=udp:eth0 children=12 Like this and the other settings, my kamailio has 21 threads and 20 Sql connections. If I add more listens, listen=udp:eth0 listen=udp:eth0:5072 listen=udp:eth0:5074 Now I have 45 threads and 45 Sql connections. Why does this happen and can i set a a number of threads and/or sql connections that doesn't change with the number of "listen" parameters that i have ? Best Regards, Duarte Rocha

3 2

recording audio stream
by bill＠telopar.net 18 Nov '18

18 Nov '18

Hi All: I have a project that requires that I record the audio streams from an ongoing phone call. It seems that rtp_proxy module should do that, but I can't seem to get it to work. My config file: *if(nat_uac_test("8")) { //rtpproxy_manage("co"); xlog("L_INFO", "rtpproxy Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwie","10.1.1.2"); } else { //rtpproxy_manage("cor"); xlog("L_INFO", "rtpproxy Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwei","10.1.1.2"); } *There appears to be conversations going on with rtpproxy on both legs of the call. But I seem to be missing the step that records the audio*. *Anybody have an example of how to do this. TIA, Bill **

4 3

Which algorithm to use as load-balance
by David Villasmil 16 Nov '18

16 Nov '18

Hello, Is it possible to set an algorithm like the following? fs1 - Priority 10 - maxload 20 calls fs2 - Priority 10 - maxload 20 calls fs3 - Priority 9 - maxload 20 calls fs4 - Priority 8 - maxload 20 calls fs5 - Priority 7 - maxload 20 calls So that, whenever any new call comes in, fs1 and fs2 would be used, meaning the fs with the highest priority would be use UNLESS there's 20 calls on that server, in which case, the next server with the highest priority would be used. Thanks!, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 7

Freeradiusclient Interim-Update Support
by Hamid Hashmi 16 Nov '18

16 Nov '18

Dear All Is module acc_radius supports interim updates? If yes then how to configure it. Regards Hamid R. Hashmi

2 3

textops module get_body_part function gives boundary error even if not multipart
by YAS0 CANER 16 Nov '18

16 Nov '18

Hello, i try to get content from Invite request but it gives boundary error. it is not multipart or mixed content but it checks boundary and gives error. Thanks. My message INVITE Content-Type: application/resource-lists+xml <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list> <entry uri="100?Call-ID=$calli&From=$fromtag&To=$toTag&Session-ID=1"/> <entry uri="103?Call-ID=$calli2&From=$fromtag2&To=$toTag2&Session-ID=2"/> </list> </resource-lists> static int ki_get_body_part_helper(sip_msg_t* msg, str* ctype, pv_spec_t *dst, int mode) { ... if(get_boundary(msg, &boundary)!=0) { LM_ERR("Cannot get boundary. Is body multipart?\n"); return -1; } .... int get_boundary(struct sip_msg* msg, str* boundary) params.s = memchr(msg->content_type->body.s, ';',msg->content_type->body.len); if (params.s == NULL) { LM_INFO("Content-Type hdr has no params <%.*s>\n", msg->content_type->body.len, msg->content_type->body.s); return -1; }

1 0

signal 15 received and .. on Kamailio 5.1.6
by AL RSM 16 Nov '18

16 Nov '18

Hi, I just installed Kamailio 5.1.6 on Ubuntu 18.04. Starting Kamailio with the default config will give us the following in the log. Is this normal? Is there a way to fix this issue? Thanks, AL Nov 15 14:01:59 lab-proxy03 kamailio[13544]: * Stopping Kamailio SIP Server: kamailio Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5182]: NOTICE: <core> [main.c:692]: handle_sigs(): Thank you for flying kamailio!!! Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5213]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5212]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received .... Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5182]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Nov 15 14:02:00 lab-proxy03 kamailio[13544]: ...done. Nov 15 14:02:00 lab-proxy03 kamailio[13556]: * Starting Kamailio SIP Server: kamailio Nov 15 14:02:00 lab-proxy03 kamailio: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Nov 15 14:02:00 lab-proxy03 kamailio: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) .... Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13618]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/13618 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13619]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Nov 15 14:02:00 lab-proxy03 kamailio[13556]: ...done.

2 1

priority load balance
by David Villasmil 15 Nov '18

15 Nov '18

Hello, Trying to do priority load balancing (8) but i'm getting an error `unable to update destination load - classic dispatching` anyone knows why? i've got ```modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)")``` Looking at the code, dstid_avp would be null, but not sure why? I have set in the table the priorities as 10 for 2 gws (edited) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

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