- sr-users - kamailio.org

cfg_get in async_worker causes segfault (v4.4)
by Malcolm O'Hare 09 Dec '17

09 Dec '17

I want to experiment with using the async module. I added the module and updated my request_route to call async_task_route() and put all the existing routing logic in a new route. Unfortunately my kamailio is crashing looking up a cfg_get variable. The exact same route logic works fine in the normal child process, its only when its run in a async_worker that it fails. Is this a known issue? I can see in the core file that there is no group handle in the cfg_group_t (its 0x0), but I know nothing about how that gets set and where. Thanks in advance for any help Malcolm

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Forwarding Registration to Asterisk
by Wilkins, Steve 07 Dec '17

07 Dec '17

Hello All, Problem: My main issue is that when I "REGISTER" a client via Kamailio, and I attempt to "Dial" a different endpoint within an Asterisk Dial Plan, Asterisk throws an error stating that the endpoint (the number I am dialing via "Dial") is not registered or reachable. However, commands like "Playback" do work correctly for the client I "REGISTERED" via Kamailio. E.g. I register client 10001 in Kamailio, I then register another client 10002 in Kamalio; both 10001 and 10002 can exercise an Asterisk Dial Plan which will play videos/audio (No Problem). But, now I want 10001 to Call (Dial) 10002; it is at this point that Asterisk throws the error "10002 is not registered or reachable". I have tried many of the suggestion on many different help boards (several times) but I am still unable to forward a registration from Kamailio to Asterisk. With my current Kamailio configuration (I do use dispatching), I see , via tcpdump, Asterisk receiving a "REGISTER" request, and Asterisk sends back the "unauthorized" as expected, however, Kamailio does not re-send the "REGISTER" as is customary. I am not sure of the next step to take, but I feel I have a couple of options. * I can continue to try and figure out why Kamailio is not sending the second "REGISTER" (I have not yet been able to figure this out). * Tell Asterisk to not require authentication. (I am using pjsip and do not know how to not require authentication in Asterisk when the request is from Kamailio). I have put a lot of time into this one, and I am at a sticking point. Any help or suggestions would be very much appreciated. Thank you,

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400 Bad From URI reply from kamalio
by Konstantin Tumalevich 07 Dec '17

07 Dec '17

One of our clients expiriences strange error. Their system sends BYE with different from address (which IP is presented in Contact header of initial INVITE) and Kamailio immediately responded with "400 Bad From URI". In which module can be generated this response? We use 4.3 version. -- Skype: konstantin.tumalevich

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Question about parsing a jsonrpc_exec result with a colon in object name with jansson_get
by jenus＠cyberchaos.nl 07 Dec '17

07 Dec '17

Hi all, I'm trying to parse the result of a json rpc call to get the register statistics. # Get registrar stats jsonrpc_exec('{"jsonrpc": "2.0", "method": "stats.get_statistics", "params": ["accepted_regs"], "id": 1}'); jansson_get("result[0]", $jsonrpl(body), "$var(registrar_accepted)"); xlog("LOG_LOCAL3","L_WARN","$jsonrpl(body)\n"); The $jsonrpl(body) var contains: {"jsonrpc":"2.0","result":["registrar:accepted_regs = 2"],"id":1} With this code the result stored in $var(registrar_accepted) is : "AcceptedRegistrations":"registrar:accepted_regs = 1" I'm trying to get only the value. I have tried: jansson_get("result[0].registrar:accepted_regs", $jsonrpl(body), "$var(registrar_accepted)"); jansson_get("result[0].['registrar:accepted_regs']", $jsonrpl(body), "$var(registrar_accepted)"); jansson_get("result[0]['registrar:accepted_regs']", $jsonrpl(body), "$var(registrar_accepted)"); But this does not work. Do i need to escape the colon in the object name "registrar:accepted_regs" ? Thanks, Jan

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Doing a quick test with http_async_client module from master branch
by Daniel-Constantin Mierla 07 Dec '17

07 Dec '17

Hello, anyone around here that can do a quick test with http_async_module? I don't have a testbed for it right now and I would spend the time on other resources if someone here can run some tests with http_async_client module from master branch. I did a bit of internal refactoring in order to be able to export the function to kemi interface -- nothing radical, but it would be good to know it didn't break anything. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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IMS with online Charging system
by Amar Tinawi 06 Dec '17

06 Dec '17

Hello all Is there any open souce ocs can be integrated with kamailio IMS ? Is it possible to integrate with CGrates ?

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t_set_fr behaviour
by Kelvin Chua 06 Dec '17

06 Dec '17

hi guys, has anyone of you tried playing around with a scenario similar to this? A. invite -> t_set_fr() 2 seconds - if "100 trying" not received in 2 seconds, timeout. works fine. B. after receiving "100 trying" received, t_set_fr() 30 seconds - if 18x not received in 30 seconds, timeout. works fine C. after receiving the first 180, t_set_fr() 10 seconds - if "200 ok" not received in 10 seconds, it will not timeout, but instead it will timeout 30 seconds after B. i noticed this behavior recently and noticed that when a previous t_set_fr() is bigger than the new one, it will be ignored. so if i did 2 -> 10 -> 30, it works (swapping timeout values of B and C) my question is, is this an expected behavior? Kelvin Chua

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load testing ims_charging
by Kelvin Chua 06 Dec '17

06 Dec '17

i am trying to load test the ims_charging module and notice some issues whenever i send test calls one by one, it's perfect. but when i use sipp and generate a bunch of calls, i would get this intermittently CRITICAL: ims_charging [ro_timer.c:122]: insert_ro_timer(): Trying to insert a bogus ro tl=0x7f207e91c2d0 tl->next=0x7f207e8b57b8 tl->prev=0x7f207e97c398 if i continue to hammer the server, it would end up failing all calls. When this issue happens, Ro_CCR will not return any value and will also not trigger the async route, looks like it just sits there frozen. during this time, sip client will receive a 408. also notice cdp complaining of latency usually around 1 second, using packet capture i am sure diameter server is responding properly. i think there is a choke point somewhere. i tried playing around with async_workers, no improvement. any pointers? Kelvin Chua

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Re: [SR-Users] Kamailio Multi IP
by Duarte Rocha 06 Dec '17

06 Dec '17

Thnaks for the reply. I don't need mhomed since i do the socket forcing by myself after the the DISPATCH selects the Gateway. What i need is something that modifies the SIP Request in order to use the outbound IP in the R-URI, To-Header, etc. I'm doing that using the $td and $rd pseudo-variables, but i'm not sure if that is the safest approach. The TOPOS module is really usefull for this, thanks for mentioning. However, the dialog_expires parameter is really confusing, since it imposes a maximum time allowed for a dialog to exist (After the records are deleted, routing no longer works). I could work with a very long timer for this if the ended dialogs would get removed from the table automatically, but that doesn't happen. Cheers

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redis via haproxy
by Aidar Kamalov 06 Dec '17

06 Dec '17

Hello, Kamailio after restart(or start) sometimes can't connect to redis (via haproxy) with logs: Dec 6 07:29:30 siptest /usr/sbin/kamailio[47291]: NOTICE: <script>: ROOT: KDMQ usrloc [764c076919c73540-39925(a)192.168.150.225] Dec 6 07:29:31 siptest /usr/sbin/kamailio[47263]: NOTICE: <script>: ROOT: OPTIONS <null> [0b4dc11225a6fc77294de97578061844@192.168.150.231:5060] Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: ndb_redis [redis_client.c:130]: redisc_init(): error communicating with redis server [redis] (127.0.0.1:6380/0): Resource temporarily unavailable Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: ndb_redis [ndb_redis_mod.c:122]: child_init(): failed to initialize redis connections Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: <core> [core/sr_module.c:923]: init_mod_child(): Error while initializing module ndb_redis (/usr/lib64/kamailio/modules/ndb_redis.so) Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: <core> [core/pt.c:334]: fork_process(): init_child failed for process 5, pid 47269, "udp receiver child=4 sock=192.168.8.213:5060" Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: CRITICAL: <core> [main.c:1611]: main_loop(): Cannot fork Dec 6 07:29:31 siptest /usr/sbin/kamailio[47252]: ALERT: <core> [main.c:740]: handle_sigs(): child process 47269 exited normally, status=255 Dec 6 07:29:31 siptest /usr/sbin/kamailio[47362]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 Redis and haproxy working fine (I test it by many attepts - echo ping|ncat localhost 6380) At tcpdump all attemps from kamailio have: *1 $4 PING +PONG *2 $6 SELECT $1 0 +OK my kamailio 5.0.4 config: modparam("ndb_redis", "server", "name=redis;addr=127.0.0.1;port=6380") my haproxy 1.5 config: backend bk_redis mode tcp option tcp-check tcp-check connect tcp-check send PING\r\n tcp-check expect string +PONG tcp-check send QUIT\r\n tcp-check expect string +OK server sipkazan 192.168.107.195:6379 check inter 1s server local 127.0.0.1:6379 check inter 1s backup -- Aydar A. Kamalov

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Re: [SR-Users] using bcrypt passwd hashing
by Walter Martín Villalba 06 Dec '17

06 Dec '17

Hello everyone, and thanks very much for your feedback. Some responses and further questions below. Daniel> Latest kamailio versions support also SHA256 algorithm Martín> SHA256 is also a bad choice for storing passwords. See details here: https://crackstation.net/hashing-security.htm Daniel> However, the main blocker in suing a different hashing algorithm are the sip client devices (mainly hardphones), which implement only MD5. If you implement your own client app, then you can extend kamailio to support whatever hashing you do in the client. Then, of course you can use client side tls certificates for authentication, which should be better than any hashing algorithm. Martín> I do implement my own client app, even though I use a third party SIP stack, which currently doesn't support any other auth methods besides basic and MD5 (standard ones). I am planning to send username and passwd as custom SIP headers in the REGISTER message, probably encrypted, and this will travel on top of TLS. Then Kamailio can extract these custom headers and call a custom python script to decrypt the values and do the authentication (bcrypt password and compare with the one in database). Client certificates are good but only in certain situations (e.g. not if you want a zero footprint client such as a web-based client), and in most cases a pain to manage when your user base grows. Alex> Do you know of any mainstream SIP UACs which support anything other than standard MD5 digest auth? Martín> I don't, but haven't really worked much at all with 3rd party SIP clients. I doubt there's any support for newer passwd hashing schemes, unfortunately. ---------- Now the details.... I'm looking at sipcomm.cfg and see it calls www_authenticate (defined in modules/auth_db/authorize.c). I believe I would need to create a similar function, e.g. bcrypt_authenticate, and call this instead, with the username and passwd values I get in my custom headers (as explained above). The routine would decrypt the values, look up the user in the database, bcrypt the passwd extracted from the custom header, and compare with the one in the database. Doesn't sound too hard, but I do have some concerns related to other functions that www_authenticate may be doing, that I would also need to do in my bcrypt_authenticate function in order to keep Kamailio functioning properly. For example, www_authenticate could be changing some values in the database and/or other temporary storage. I took a quick look at the implementation and tried to follow the calls inside it. I see calls to mark_authorized_cred, check_auth_hr (or auth_check_hdr_md5), and generate_avps, and that some of these functions are indeed changing some values here and there. So, before spending more time looking into these details, I wanted to see if any of you have any suggestions about how to handle this situation, i.e. maybe all I need to do in bcrypt_authenticate is to check the credentials and then set one flag in the database for the user that was just authenticated? Does the explanation above make sense to you? Please let me know any suggestions or further guidance you may have. Thanks a lot, Martín. On Mon, Nov 13, 2017 at 3:00 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. Branch 5.1 created (Daniel-Constantin Mierla) > 2. Development open in master branch (to be v5.2.x) > (Daniel-Constantin Mierla) > 3. Re: using bcrypt passwd hashing (Daniel-Constantin Mierla) > 4. Re: t_set_fr behaviour (Daniel-Constantin Mierla) > 5. Re: t_set_fr behaviour (Daniel-Constantin Mierla) > 6. Re: AVPOPS: is_avp_set/avp_check "name" parameter as > variable. (Daniel-Constantin Mierla) > 7. Re: strange --dialog in delete state is too old-- log line > managing dialog hashes (Daniel-Constantin Mierla) > 8. Re: 183 acc records even if early_media equals to 0 > (Marco Capetta) > 9. Re: Kamailio issue (Daniel-Constantin Mierla) > 10. Re: using bcrypt passwd hashing (Yuriy Gorlichenko) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 12 Nov 2017 14:42:35 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org>, > "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org > > > Subject: [SR-Users] Branch 5.1 created > Message-ID: <dca81faa-dec2-4e12-704f-b382d23493d7(a)gmail.com> > Content-Type: text/plain; charset=utf-8 > > Hello, > > the GIT branch 5.1 has just been created, it will host the release > series 5.1.x. To get this branch from GIT, you can use: > > > git clone https://github.com/kamailio/kamailio.git kamailio > cd kamailio > git checkout -b 5.1 origin/5.1 > > > Hopefully in two-three weeks time frame the full release of 5.1.0 will > be out. > > >From now on, any corresponding fix has to be pushed first to master > branch and then cherry-picked to branch 5.1. No new features can get in > branch 5.1. Enhancements to documentation or helping tools, as well as > kemi exports are still allowed. If you are not sure about doing or not a > backport, ask on sr-dev mailing list. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > > > ------------------------------ > > Message: 2 > Date: Sun, 12 Nov 2017 14:50:45 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org>, > "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org > > > Subject: [SR-Users] Development open in master branch (to be v5.2.x) > Message-ID: <07baf03f-0d1b-30f6-45d2-cacfc3dfec99(a)gmail.com> > Content-Type: text/plain; charset=utf-8 > > Hello, > > git branch 5.1 was just created (to host the release series v5.1.x), > therefore new features can now be pushed again in master branch. They > will be part of the next future release, likely to be numbered 5.2.x. > > Any fixes that affect existing code in branches 5.1 or older version > have to be backported - push first to master and then cherry pick -- see > the contributing guidelines at: > > - > https://www.kamailio.org/wiki/devel/git-commit-guidelines# > backporting_commits > > Many thanks to all contributors so far! Testing of branch 5.1 and giving > feedback for it is very appreciated! > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > > > > ------------------------------ > > Message: 3 > Date: Mon, 13 Nov 2017 09:22:17 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, Yuriy Gorlichenko < > ovoshlook(a)gmail.com> > Subject: Re: [SR-Users] using bcrypt passwd hashing > Message-ID: <c7bb57e5-16dd-f5c2-f4ac-e3060f3b45bb(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > > > On 12.11.17 10:33, Yuriy Gorlichenko wrote: > > You can realize any of auth methods by yourself and include it via > > config file/kemi on lua/by adding module > > > > forexample I added SSO auth without any troubles instead of basid MD5 > > for some projects. > Out of curiosity, what do you refer by SSO? > > Cheers, > Daniel > > > > 2017-11-11 18:49 GMT+03:00 Alex Balashov <abalashov(a)evaristesys.com > > <mailto:abalashov@evaristesys.com>>: > > > > Do you know of any mainstream SIP UACs which support anything > > other than standard MD5 digest auth? > > > > On November 10, 2017 7:11:26 PM EST, "Walter Martín Villalba" > > <wvillalba(a)gmail.com <mailto:wvillalba@gmail.com>> wrote: > > >Hello, > > > > > >I did some searches online and talked to some colleagues and it > seems > > >Kamailio only supports the traditional HTTP digest authentication, > > >which > > >uses MD5. I would like to know if any of you has been successful in > > >using > > >bcrypt/scrypt/pbkdf2 passwd hashing, instead of MD5, which has been > > >deemed > > >as obsolete and insecure a long time ago. Perhaps you've written > your > > >own > > >auth module, or just modified the config script to call some other > > >credential checking routine using a custom python/perl script (I'm > > >thinking > > >of doing the latter, of nothing better is available). > > > > > >If any of you have done something like this, using bcrypt or any > > other > > >current and secure hashing algorithm, I would appreciate some > > guidance. > > > If > > >you haven't, aren't you concerned about storing MD5 password > > hashes in > > >your > > >database? > > > > > >Note: if I can't find a good answer using this list, I will try the > > >developer's list next. > > > > > >Thanks in advance, > > > > > >Martín. > > > > > > -- Alex > > > > -- > > Sent via mobile, please forgive typos and brevity. > > > > _______________________________________________ > > 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 > > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> > > > > > > > > > > _______________________________________________ > > 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.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > >

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Kamailio Multi IP
by Duarte Rocha 05 Dec '17

05 Dec '17

Greetings, I have a Kamailio that is hosted on multiple IP's. Each client connects to an IP and there is another IP that communicates with the PSTN. The problem is that I don't want any client "seeing" any IP besides his. If the client (connected to kamailio's IP 1.1.1.1) receives an request coming from the PSTN (connected to Kamailio's IP 2.2.2.2) , it will have 2.2.2.2 on the R-URI, To Header,etc. Is there a method to ensure kamailio will manipulate the packet and use the correct IP ? Currently i'm directly changing the R-URI and To Header, but I don't if it's the correct approach to this. I would also like to know the best way to do those changes right before the packet leaves Kamailio. I know that OnSend route doesn't allow changes in the buffer. I tried Branch routes but not every request goes through that. Thanks for the help. Cheers.

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how to escape asterisk in pattern matching
by voipspace voipspace 05 Dec '17

05 Dec '17

how to escape asterisk in pattern matching if($rU=~"\d+*") regards

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How to change To header URI from Kamailio?
by Arun NV 05 Dec '17

05 Dec '17

Hi, I am using *Kamailio 4.4*. I would like to forward the request to a different port number of my endpoint. I have changed the destination *URI *and the *INVITE *correctly reached to the new port. But the *To header *in the *INVITE *request has the old port. so the endpoint is not responding to the request. Then I tried to remove and replace *To header* using *remove_hf("To")* and* insert_hf("To: $var(modified_to_header) \r\n");* functions. But the to header not changed. So, is there any way to change the *To header URI*.

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Dialog Module Hash Size vs Number of Dialogs
by Mark Blackford 04 Dec '17

04 Dec '17

Hello, I am trying to properly size the use of the Dialog Module hash for our implementation using: modparam("dialog", "hash_size", <number that is power of two>) However, in my testing, I have been unable to figure out the relationship between the hash size and a number of dialogs I need to support. I think the hash size is specifying a memory block in kB, but even setting the hash size to a very tiny number does stop me from creating hundreds of dialogs. Is there a way to determine a relationship between the hash size and a rough number of dialogs that would be expected? An example of a a dialog looks like this from kamctl: [root@kamailio01 ~]# kamctl dialog show dialog memory records dialog:: hash=22:70 state:: 4 ref_count:: 2 timestart:: 1512151205 timeout:: 36083666 callid:: 0gQAAC8WAAACBAAALxYAAClws2wyL8GE+CSgRY7HIhmg9ZUIISZad46ntOPng3iPIcLaxzLFaytRTI7M0Bzz0g--(a)10.155.8.40 from_uri:: sip:b53667d44239457fbc94fc2f4c4e25a6@sip.dcs-staging.net from_tag:: 10.155.8.40+1+689d7e5e+8fcf481a caller_contact:: sip:43f0ae1480846185e8803f21e9f2b721@10.155.8.40:5060 ;transport=udp caller_cseq:: 24115 caller_route_set:: caller_bind_addr:: udp:10.155.8.11:5060 callee_bind_addr:: udp:10.155.8.11:5060 to_uri:: sip:2052773090@sip.dcs-staging.net to_tag:: sip+1+bdcd0004+2038f37c callee_contact:: sip:ca2013e84f10348a1cc825c12562bde7@10.155.8.40:5060 ;transport=udp callee_cseq:: 0 callee_route_set:: Thanks! -- Mark Blackford Digium Cloud Services 678.230.8769

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How to use ruri-pattern with allow_trusted?
by voipspace voipspace 04 Dec '17

04 Dec '17

Hi Have used allow_trused to pass the allowed ip. Now I want to check the request has a correct tech prefix. With an entry in ruri-patten column in the trusted table for example 75622#. So if calls are to olny be allowed when the "to" destination starts with 75622#. Regards

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max_contacts does not work as expected
by Gholamreza Sabery 04 Dec '17

04 Dec '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

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Dialog Module Hash Size vs Number of Dialogs
by Mark Blackford 04 Dec '17

04 Dec '17

Hello, I am trying to properly size the use of the Dialog Module hash table for Kamailio 4.1.3 implementation using: modparam("dialog", "hash_size", <number that is power of two>) However, in my testing, I have been unable to figure out the relationship between the hash size and a number of dialogs I need to support. I think the hash size is specifying a memory block in kB, but even setting the hash size to a very tiny number does stop me from creating hundreds of dialogs. Is there a way to determine a relationship between the hash size and a rough number of dialogs that would be expected? Thanks, -- Mark Blackford Digium Cloud Services 678.230.8769

2 1

How to get call on an extension, which is registered when a call towards it reach kamailio. How to modify pending session parameters in kamailio?
by Arun NV 04 Dec '17

04 Dec '17

Hi, I am using *Kamailio 4.4* as the proxy with my *Asteri**sk *server. I am trying to develop a scenario where an extension gets registered on asterisk via *Kamailio *when it receives a push notification. This push notification is sent to the sip extension when a call towards this extension reaches to the *Kamailio*. For example, suppose there is two SIP extension(* extension 1 and extension 2*) registered on *Asterisk *via Kamailio. When a call from *extension 1* reaches the asterisk, it forwards the *INVITE *request towards *extension 2* via Kamailio.Kamailio will try to forward it to extension 2. suppose the *extension 2* is not able to receive the *INVITE *request from *Kamailio*. When extension two receive a push notification, it will register on asterisk. So I need to get the call on *extension 2 *through the new registration. We are trying to simulate registration of extension to the asterisk when receiving the push notification. First, we registered *extension 2 *and disconnected the network. Then we tried to register the same extension when a call from *extension 1 *reaches to Kamailio. This is a simulation of push-based registration since an extension that receives the push will attempt to register when an incoming call is received. When asterisk sends *INVITE *request to Kamailio, it immediately responded with *100 trying* provisional response. This *100* response by Kamailio towards asterisk prevents asterisk from re-transmitting the INVITE. Then Kamailio tried to send and retransmit the packet to *extension 2*, which does not have network access. This* extension 2 *was on port number *24071*. Even after successful registration(in port *59995*) of the *extension 2*, Kamailio continued to transmit the packets to the old port. After that, we have configured Kamailio in a way that it won't send an immeadeate provisional reply(*100 trying* ) for *INVITE *request. Here Kamailio is not immediately sending *100 trying* message to *Asterisk*. This forces Asterisk to re-transmit. Asterisk was found to retransmit the same packets. However, even after the successful registration of* extension 2*, asterisk continued to send the old invite to Kamailio not the new one to the latest port. This is the problem for me since push relies on the *INVITE *reaching the phone at the correct port number. So, is there other good approaches to solve this issue? One thing I would like to try is modifying the pending *INVITE *request towards old registered port with the new port details when new registration reaches to Kamailio. Can I get the ongoing requests from Kamailio? Please suggest a viable solution. Thank you, Arunbalan

2 1

Using dispatcher's algorithm 12
by George Diamantopoulos 04 Dec '17

04 Dec '17

Hello all, I've been trying to use dispatcher's algorithm "12" to fork REGISTER messages constructed with uac module to all destinations within a set. However, the request is routed to only one destination. Is anything else needed in addition to calling ds_select_dst("0", "12") in the configuration script? Thanks. BR, George

3 5

bash script to install siremis&kamailio
by Atux Atux 04 Dec '17

04 Dec '17

Hi. i would like to ask if someone has found any bash script that could install all at once for Kamailio and Siremis in ubuntu/debian or even in centos.

2 1

dmq issues
by Aidar Kamalov 03 Dec '17

03 Dec '17

Hello. I have 4 kamailio nodes with same config: modparam("dmq", "server_address", "sip:LOCALIP:5050") modparam("dmq", "notification_address", "sip:192.168.8.213:5050") modparam("dmq", "notification_address", "sip:192.168.10.54:5050") modparam("dmq", "notification_address", "sip:192.168.107.195:5050") modparam("dmq", "notification_address", "sip:192.168.150.225:5050") modparam("dmq", "num_workers", 6) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_size", 2000) modparam("dmq_usrloc", "batch_usleep", 1000) regularly (don't know why) I've got issues - replication is broken, for example now at all nodes I have: # kamcmd dmq.list_nodes { host: 192.168.8.213 port: 5050 resolved_ip: 192.168.8.213 status: 2 last_notification: 0 local: 0 } { host: 192.168.10.54 port: 5050 resolved_ip: 192.168.10.54 status: 2 last_notification: 0 local: 0 } { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 8 last_notification: 0 local: 0 } { host: 192.168.107.195 port: 5050 resolved_ip: 192.168.107.195 status: 2 last_notification: 0 local: 1 } but at node 192.168.150.225: # kamcmd dmq.list_nodes { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 2 last_notification: 0 local: 0 } { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 2 last_notification: 0 local: 1 } nodes restart doesn't help. but if i exec systemctl stop kamailio && sleep 15 && systemctl start kamailio it is partially solve my issue - dmq.list_nodes show all nodes, but at 192.168.150.225 i see not all user locations: they are added gradually There is no errors in logs Dec 1 08:28:37 sipufa /usr/sbin/kamailio[18252]: NOTICE: <script>: ROOT: KDMQ notification_peer [15f3edf9258014d1-7114(a)192.168.8.213] Dec 1 08:28:38 sipufa /usr/sbin/kamailio[18271]: NOTICE: <script>: ROOT: KDMQ notification_peer [7acb9eae627482f8-87828(a)192.168.10.54] Dec 1 08:28:39 sipufa /usr/sbin/kamailio[18250]: NOTICE: <script>: ROOT: KDMQ notification_peer [621ad9f474b335e2-18272(a)192.168.150.225] Dec 1 08:28:40 sipufa /usr/sbin/kamailio[18252]: NOTICE: <script>: ROOT: KDMQ notification_peer [7b817f6a6244f923-8872(a)192.168.107.195] Dec 1 08:28:47 sipufa /usr/sbin/kamailio[18271]: NOTICE: <script>: ROOT: KDMQ usrloc [7acb9eae627482f2-87794(a)192.168.10.54] Dec 1 08:29:20 sipufa /usr/sbin/kamailio[18251]: NOTICE: <script>: ROOT: KDMQ usrloc [7acb9eae627482ef-87790(a)192.168.10.54] -- Aydar A. Kamalov

3 8

kamailio as application server
by eyas barhouk 01 Dec '17

01 Dec '17

Hello if i need to use kamailio as an application server like mmtel for example how could i do that and what is the features that kamailio have if i want to use it as application server in another word what is the type of application server it could be act like (mmtel - presence) ? and thanks for help

2 1

In need of a simple, stateless 302 redirect for INVITE
by Robert 01 Dec '17

01 Dec '17

Hello again, I am not a SIP expert, and I’m new to Kamailio, really enjoying learning it, but I’ve hit a blocker and I’m pretty sure I’ve missed something really simple. I’m looking to implement the Redirect Server from here: https://tools.ietf.org/html/rfc3665#section-3.6 <https://tools.ietf.org/html/rfc3665#section-3.6> (the Avaya Session Manager being Alice), Kamailio as the Redirect Server. My configuration is working fine, and I can see the 302 going back (to Avaya SM), I then get the ACK I’m expecting. The incoming SIP request contains a number of fields, including a Contact header, which I’m told SM needs on the return. But the send_reply() or sl_send_reply() doesn’t return the Contact header, and I am unable to manually set it, or any other header, using any of the append_hf(), append_to_reply() etc. mechanisms. I’m executing a msg_apply_changes() which succeeds, but no Contact header is returned. I’ve tried an sl_send_reply(“100”, “Trying”) before any other processing, and Wireshark confirms that the SIP response is basically identical to the 302 response sent after processing… The goal is to run this stateless, I’m assuming this is possible - it certainly feels like it should be. I’ve re-read the intros & tutorials, and other reading material such as this http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf <http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf> but, this, even my best Google-fu has left me utterly stumped… The sl_send_reply documentation https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply <https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply> states “If the code is in the range 300-399 (redirect reply), the current destination set is appended to the reply as Contact headers” but the Contact header is definitely not being set when I use this. Apologies in advance if eyes are rolling and this is really obvious, or (and?) that my lack of SIP knowledge is shining through - any tips or guidance would be appreciated. I certainly feel overwhelmed by the number of modules and the interplay between them. If the answers to this, and other things along these lines are contained in the admin book, that’s fab, I’ve just filled the form out :) Cheers - Robert...

3 3

DBURL password in clear
by Robert 01 Dec '17

01 Dec '17

Hello, I’m working for a UK high street bank and our Kamailio implementation has been challenged because we’ve got database passwords held in clear in the configuration file. I am unable to find any examples of where this has been worked around, there doesn’t seem to be any module or configuration means of supplying a variable in the modparam() entry that is expanded a startup. The security tutorials only seem to relate to the SIP level of security, not Kamailio as a platform. My requirement is simple, I need to be able to supply a password via means such as loading a variable from a run-once script at start up, or a module. The ideal would be to be able to read in a Docker secret :) I am by no means a Kamailio expert, so apologies in advance if this is a mindblowingly basic thing to achieve, but I do feel I’ve exhausted the Kamailio documentation, wiki etc. and all the goodness Google usually has to offer and drawn a blank. Sincere thanks in advance for any assistance. Cheers - Robert...

5 14

Dispatcher latency stats monitoring with statsd
by Julien Chavanton 01 Dec '17

01 Dec '17

Not sure if anyone is using "latency measurement" feature in Kamailio dispatcher module, I thought I could share a very simple Python script that I am using to monitor remote gateway using this. Sample Script is attached, it will post data to statsd and is working in cron The feature can be enabled : 3.27. ds_ping_latency_stats (int) Enable latency measurement when pinging nodes If set to 0, disable latency measurement. If set to 1, enable latency measurement https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc… This is one use case of this feature, another one will be to load balance based on congestion detected on latency, I am planning to share a lab experiment in a little while. Regards

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 30 Nov '17

30 Nov '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 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, - related events around FOSDEM, including the XMPP summit, - 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, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc 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 30th of November. 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/FOSDEM18/ 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/2018/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, 3rd 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 Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd 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: https://lists.fsfe.org/mailman/listinfo/free-rtc 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: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

3 2

blf in kamailio dmq cluster
by Aidar Kamalov 30 Nov '17

30 Nov '17

Hello, I have two(or more) kamailio servers with dmq module and dmq_usrloc - sip1.example.com and sip2.example.com publish and subcribe replication works too. BLF works fine if clients connected to the same kamailio server, because in active_watchers table stored with same domain sip1.example.com But what to do when call going from one server to second? Is there any configuration to ignore domain in pusblish, subscribe, notify, etc? -- Aydar A. Kamalov

3 7

OpenSIPS-devel not starting (TCP Error)
by SamyGo 29 Nov '17

29 Nov '17

Hi, I just installed the head version from git. version: opensips 2.4.0-dev (x86_64/linux) flags: STATS: On, SHM_EXTRA_STATS, EXTRA_DEBUG, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: d1a4419 main.c compiled on 19:53:20 Nov 28 2017 with gcc 4.8 It was compiled with EXTRA_DEBUG and DBG_TCPCON flags; While starting opensips with the residential vanilla script it gives following error: Nov 28 20:05:52 [30899] ERROR:core:tcp_init: oom con hist Nov 28 20:05:52 [30899] CRITICAL:core:main: could not initialize tcp Nov 28 20:05:52 [30899] INFO:core:cleanup: cleanup Nov 28 20:05:52 [30899] DBG:core:shm_mem_destroy: destroying the shared memory lock Nov 28 20:05:52 [30899] NOTICE:core:main: Exiting.... already running Even if I remove the proto_tcp, and tcp related things this still gives error and doesn't start. Kindly help. Regards, Sammy

2 3

parse_from_header(): bad From header
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. We experience the problem with malformed messages. First of all, I want to say, that most of time kamailio works well and nothing happens that can drop sessions. But from time-to-time something changes (may be in the network) and kamailio receives requests with malformed headers (To or From hfs). The schema: uplinks -> kamailio <-> routing server (asterisk) Malformed messages were received only from asterisk server. Sip debug on asterisk showed that messages were transmitted correctly and headers were nice. But kamailio obtains changed (broken) packets or perhaps can't read them properly. How does it look like: Nov 10 12:37:06 kamailio-name kamailio[965]: INFO: <script>: Going to NATMANAGE for BYE from sip:useragent7122@kamailio_address:5068 (IP=wss:client_address:62881) - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: NOTICE: <script>: Relaying request to <null> - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: uac [replace.c:783]: restore_uris_reply(): failed to find/parse FROM hdr Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: Skip manage DEVICE_STATE for BYE from <null> (IP=udp:asterisk_address:50600) with (200 - OK) - R=<null> ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: ------------------------------------MANAGE BYE by DEVICE_STATE_BYE_MANAGE route - 200 - OK M=BYE IP=udp:asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine_funcs.c:331]: get_from_tag(): failed to parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine.c:2252]: rtpp_function_call(): can't get From tag Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header How the packet looks like: ACK sip:tt7etmau@bb0pd5t63b26.invalid;alias=client_address~57573~6;transport=ws SIP/2.0 Via: SIP/2.0/UDP asterisk_address:50600;branch=z9hG4bK117106ca;rport Route: <sip:kam2.domain.com:5068 ;nat=yes;transport=udp;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331>,<sip:kam2.callision.com:5068 ;nat=yes;transport=ws;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331> Max-Forwards: 70 From: "19172423539" <sip:19172423539@asterisk_address:50600>;tag=as06869a5d To: <sip:useragent7122@10.0.1.18:5068>;tag=nmv7kmpau3 Contact: <sip:19172423539@asterisk_address:50600> Call-ID: 629d5a132c194c536b5f5c1a2a3c6e32@asterisk_address:50600 CSeq: 102 ACK User-Agent: asterisk PBX Content-Length: 0 The transport is used to send messages between asterisk and kamailio is TCP. TCP configurations: tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 I found the idea, that we need to change following parameters: modparam("uac","restore_mode","auto") modparam("uac","restore_dlg",1) to: modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) I did that, now it looks like all is fine, but I think it can get back. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua>

3 5

Source socket/port for in-dialog requests
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. My apologies for so frequent appealing to you. I'm trying to solve problem with ending of sessions. The problem consists of no 200 OK coming from uplink on our BYE requests. Topology. First leg: Webrtc client <-wss-> kamailio <-sip tcp-> asterisk routing server Second leg: Uplink <-sip tcp-> kamailio <-sip tcp-> asterisk routing server The problem appears only in case when dialog was ended by webrtc client. The first leg (dialog) of the call ends nice, without any hints on problem. But the second leg (with uplink) has problem with no 200 OK (coming from uplink) on BYE request coming from asterisk. Handshake between asterisk server and uplink establishes properly. It looks like: uplink.domain.com:5060 -> INVITE tcp -> our.kamailio.server:5060 -> INVITE tcp -> our.asterisk.server:5060 . uplink.domain.com:5060 <- 100 trying tcp <- our.kamailio.server:5060 . uplink.domain.com:5060 <- 180 ringing tcp <- our.kamailio.server:5060 <- 180 ringing tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 <- 200 OK tcp <- our.kamailio.server:5060 <- 200 OK tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 -> ACK tcp -> our.kamailio.server:5060 -> ACK tcp -> our.asterisk.server:5060 . uplink.domain.com <-media stream-> rtpengine <-media stream-> our.asterisk.server . Here kamailio starts to use random source port for relaying in-dialog BYE uplink.domain.com:5060 <- BYE tcp <- our.kamailio.server:45355 <- BYE tcp <- our.asterisk.server:5060 . And here the leg with uplink is expected to end with 200 OK (coming from uplink proxy). But uplink doesn't answer at all. We requested a tcpdump from uplink to see how packets are forwared from their side. And I saw that that 200 OK tries to be sent within first tcp session - to 5060 port of kamailio server. But sngrep on our side shows nothing, nothing appears in kamailio log, so 200 OK can't reach the 5060 socket because of some transport problem I think. Top via hf of our BYE request, contains kamailio record = uplink.domain.com:5060 My main thought on this count is to suppress using of random source ports for in-dialog requests, this behaviour looks irrelevant. We have turned on mhomed parameter (mhomed=1). And cookbook says: "Set the server to try to locate outbound interface on multihomed host. This parameter affects the selection of the outgoing socket for forwarding requests." But, there is a big but - we can not turn of this parameter for this moment, because routing script made with using of this one. Kamilio works on AWS EC2 platform. It has following IP address schema: private address atouched to container -> public address assinged to this private address (so this is standard ip address schema for AWS containers). For this example private address will be: 10.0.0.1 and public address will be: 100.0.0.1 Configuration - main parameters (all values are changed for an example): advertised_address=our.kamailio.server advertised_port=5060 alias=our.kamailio.server alias=our.kamailio.server:5060 alias=100.0.0.1 alias=100.0.0.1:5060 alias=10.0.0.1 alias=10.0.0.1:5060 auto_aliases=no port=5060 listen=100.0.0.1 listen=10.0.0.1 listen=tcp:100.0.0.1:5060 listen=tcp:10.0.0.1:5060 mhomed=1 fork=yes fork_delay=5000 children=6 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 One of the solutions to this problem was to add following inside the relaying route block: $fs = "tcp:" + "10.0.0.1" + ":5060"; if (!t_relay()) { sl_reply_error(); } But it seems to me it's not a smart solution. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 1

Best way to do routing against a list of gateways
by voipspace voipspace 29 Nov '17

29 Nov '17

3 6

allow_trusted not passing
by voipspace voipspace 29 Nov '17

29 Nov '17

4 5

URL problems connecting to external (Azure) MySQL database
by Floimair Florian 29 Nov '17

29 Nov '17

Hi! I'm having troubles connecting Kamailio to an external MySQL database (on Microsoft Azure Cloud). Initially creating the database with kamctl worked fine, but accessing it at runtime via db_url fails. A somewhat unusual thing about the connection to a MySQL database on Azure is that the username contains an '@' character. This leads me to believe that this case is somewhat unhandled when parsing the database URI. Trying to connect to something like modparam("sipcapture", "db_url", "mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db") Results in errors starting Kamailio ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'clouddb' (-2) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: sipcapture [sipcapture.c:1146]: child_init(): unable to connect to database [mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db] from capture_mode param. Is there a trick on how to achieve a connection in this case like some sort of escape character probably? Mit freundlichen Grüßen Florian Floimair (IMS) 1568

2 4

lcr_id vs lcr_count
by Patrick Wakano 29 Nov '17

29 Nov '17

Hello Kamailio list! I've been working with the LCR module and I wanted to use the lcr_id to match some ID's I have somewhere else. However, due to the lcr_count my lcr_ids must be less than the lcr_count. I understand the reason behind the lcr_count, what I think is a mistake is to have the lcr_id dependent of the lcr_count. Ideally we should be able to use any lcr_id value, just observing that the total amount of lcr groups should not exceed the lcr_count. The problem is this if that appears in the load_gws, to_gw and from_gw functions in lcr_mod.c: if ((lcr_id < 1) || (lcr_id > lcr_count_param)) { LM_ERR("invalid lcr_id parameter value %d\n", lcr_id); return -1; } I ended up using another table to make the conversion of ids, but this is horrible.... Any other reason to force the lcr_id to be less than the lcr_count? Best Regards, Patrick Wakano

3 3

how to add new media server in dispatcher table through "ANSIBLE"
by Narayan P 29 Nov '17

29 Nov '17

Hello Experts, I am new to ansible and deploying kamailio through ansible.Sucessfully deployed kamailio but my requirement is to add media server through ansible.As I am using dispatcher module in kamailio,not able to find any script that directly insert one media server IP and port into dispatcher table from ansible.(I am using ansible in one system and deploying kamailio in other). Any suggestion will be highly appreciated as always. Thanks, Narayan +91-9035438928

3 4

Re: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue.
by otron2016＠gmail.com 29 Nov '17

29 Nov '17

Through Firefox likely means via websockets (wss) rather than via the sip port. Many things could interfere with the latter. Take heart, though: websockets is harder to set up so you're likely through the worst. Look at logs and see what iftop shows you about traffic on the server as you try to make sip connections. Also try your sip client with a known-good sip server (e.g. Iptel) just to make sure the problem isn't with the client. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/28/2017 6:39 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue. Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won’t work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don’t know 100% for sure. Thank you

1 0

Using SIP tools through Kamailio and Asterisk - chrome issue.
by Wilkins, Steve 28 Nov '17

28 Nov '17

Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won't work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don't know 100% for sure. Thank you

1 0

dmq replicate
by Aidar Kamalov 28 Nov '17

28 Nov '17

Hello, I set up kamailio cluster. dmq_usrloc works fine, but PUSBLISH and SUBSCRIBE replication not. If I make a call to first node I've got many(hundreds) PUBLISH messages from first node to first node and some messages (8 for example) to other nodes. And second xnotice in config not executes. And errors n log ERROR: <core> [core/sr_module.c:1726]: get_int_fparam(): Unsupported param type for int value: 32 How to replicate publish and subscribe, is it wrong way? my confiig: request_route { if(is_method("KDMQ")) { if ($Rp == 5050) { dmq_handle_message(); exit; } else { drop; } } ..... ..... if(is_method("PUBLISH|SUBSCRIBE")) { if(!dmq_is_from_node()) { xnotice("AAAAAAAA"); dmq_t_replicate("1"); xnotice("AAAAAAAABBBBBBBBBBB"); } } 2017/11/28 12:56:06.415601 192.168.8.213:5060 -> 192.168.8.213:5060 PUBLISH sip:34200@siptest. SIP/2.0 Via: SIP/2.0/UDP 192.168.8.213;branch=z9hG4bKc7a5.cee84051000000000000000000000000.0 To: <sip:34200@siptest> From: <sip:34200@siptest.>;tag=ae81df16b5a52fc1f62f9b45f337e660-db10 CSeq: 10 PUBLISH Call-ID: 4c8f9e6e799efcd7-57244(a)192.168.8.213 Content-Length: 556 User-Agent: kamailio (5.0.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Expires: 125 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:34200@siptest."> <dialog id="898932197-5060-86(a)BJC.BGI.D.DB" call-id="898932197-5060-86(a)BJC.BGI.D.DB" direction="initiator"> <state>Trying</state> <remote> <identity>sip:53003@siptest.t</identity> <target uri="sip:53003@siptest.tg"/> </remote> <local> <identity>sip:34200@siptest</identity> <target uri="sip:34200@siptest"/> </local> </dialog> </dialog-info> -- Aydar A. Kamalov

1 1

dispatcher/carrierroute and branches across SBC and remote peer
by Nathan Ward 28 Nov '17

28 Nov '17

Hi, We use Kamailio as a SIP server, and SEMS as an SBC. We have a need to use the dispatcher/carrierroute modules to provide HA/load sharing across two different parts of our environment: 1) Our SBCs 2) Our peer’s SIP servers We have pairs of SBCs (for now) and they have the same network connectivity - different IPs of course, but they can reach the same things. Our peer’s network has 2 SIP servers (again, for now) reachable through either SBC. We select which SBC pair to use based on where the calls are going, in a branch route. In some cases, we use parallel forking, in others we use serial forking (i.e. to a media server for a prompt, then sending the call to wherever it needs to go). Right now there is no automatic failover or load sharing, so this is all simple logic - if call is to x peer or y customer, use x SBC or y SBC. I’m trying to add the failover/loadbalancing now. I have a few half baked ideas about how to achieve a network like this, but thought I’d ask if anyone else had done this first before I went too overboard. Currently I’m thinking of calling cr_route to figure out which peer SIP server, from there figuring out which SBC pair, and then calling ds_select_dst to select which SBC in the pair to set $du. I’m being tripped up though, as I can’t call cr_route or ds_select_dst in a branch route, but, as far as I’m aware that’s where I’d need to do so in order to handle this for multiple branches. Should I be looping the request through the SIP server to achieve this (i.e. force $du = 127.0.0.1 in my branch_route), or, can I avoid that complexity? -- Nathan Ward

1 0

Kamailio didn't start before increasing fork_delay
by igor.potjevlesch＠gmail.com 27 Nov '17

27 Nov '17

Hello, We suddenly had an issue on one Kamailio instance: we were not able to restart. Kamailio started to boot and fork and suddenly crashed. The last logs reported a failure regarding the ability to connect to one MySQL instance. I finally succeed to restart after increasing: fork_delay=5000 to fork_delay=9000. How this could happen suddenly? We already restarted Kamailio on this server. Regards, Igor.

2 5

Does Kamailio support RFC4117?
by Ye, Charlie (NSB - CN/Qingdao) 27 Nov '17

27 Nov '17

Hi Friends, Does Kamailio support RFC4117? Do you have any suggestions if I want support RFC4117 through Kamailio? Thanks and Best Regards, Charlie

2 2

Dispatcher with kam & fs behind nat
by Daniel Greenwald 27 Nov '17

27 Nov '17

Dispatcher is sending inbound calls to Freeswitch boxes. Both the FS boxes and Kam are behind the same nat. What is the correct way to pass calls to the FS boxes so that all SIP between FS & Kam use local IP's? Should Kam put two record-route headers on, one with PUBLICIP and one with PRIVATEIP? How is this accomplished? Thanks

2 3

Kamailio 5.0 Android Calling Not working
by Zeeshan M 25 Nov '17

25 Nov '17

Hello Friends, I had successfully installed and configured Kamailio 5.0 on linux machine. And tried it with linphone app & following are the issues i am facing, 1) IOS to Android Calling not working with timeout error on server end when we checked logs 2) Android to Android calls also not working same error. Kindly let me know what could be the issue. Regards, Zeeshan.

1 0

Re: [SR-Users] sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

I just verified that I do have logging in all pertinent areas. I added tls and I do see "TLS accept successful" in the kamailio logs, however, I see no log being generated in route[xhttp:request], and I do have logging in the first line of this route. I don't think it is getting there. Also, I noticed I have event_route[xhttp:request] not route[xhttp:request]. Thank you, -Steve

1 0

sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen's in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve

2 5

Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

Just a guess but maybe later entries [like +HIGH:+MEDIUM:+LOW] put it back. Try switching the order so that !ECDHE and the others you're trying to exclude come after. Sent from Samsung Mobile -------- Original message -------- From: Ilyas Keskin <ilyask92(a)gmx.de> Date: 11/24/2017 10:19 AM (GMT-08:00) To: miconda(a)gmail.com,"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite Hi Daniel, yes I am using the tls.cfg file. I tried your suggestion to add the cipher suite string (notice the !EDCHE which I also added to the httpd ssl.conf) but nothing changed. [server:default] method = TLSv1 cipher_list = !DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem certificate = /etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem #ca_list = ./modules/tls/cacert.pem #crl = ./modules/tls/crl.pem Also here is a log snippet from tls module section of kamailio initialization. Notice first two lines. Also it seems to me the module actually ignores the local openssl installation and uses its own which has been compiled with the module itself (?). Other than that it seems to be accepting the cipher_list value just fine: Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:355]: mod_init(): With ECDH-Support! Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:358]: mod_init(): With Diffie Hellman Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:587]: init_tls_h(): tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, compression: on Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:595]: init_tls_h(): tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, zlib compression: compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_ Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tls [tls_init.c:649]: init_tls_h(): tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 7864320 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 3932160 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_branch("MANAGE_BRANCH"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_reply("MANAGE_REPLY"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_failure("MANAGE_FAILURE"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSs<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSs<default>: certificate='/etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSs<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSs<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSs<default>: require_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSs<default>: cipher_list='!DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSs<default>: private_key='/etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSs<default>: verify_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSs<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:670]: set_verification(): TLSs<default>: No client certificate required and no checks performed Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSc<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSc<default>: certificate='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSc<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSc<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSc<default>: require_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSc<default>: cipher_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSc<default>: private_key='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSc<default>: verify_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSc<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:655]: set_verification(): TLSc<default>: Server MUST present valid certificate Would it be possible to compile the tls module with certain openssl config switches (i.e. no-ec no-dh)? Any other ideas? Best regards, Ilyas Keskin Am 24.11.2017 um 15:45 schrieb Daniel-Constantin Mierla: Hello, On 23.11.17 22:42, Ilyas Keskin wrote: Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. are you also using tls.cfg? If yes, there is an attribute for chiper list in it as well, try and see if works with it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Re: [SR-Users] sipML5 through kamailio
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

"So, tls is required for wss?" Yes, as of the last rfc. It can work without it if the client software (browser) will accept it though by now most will treat it as insecure. I forget whether sipML5 will but my guess would be no. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/24/2017 9:57 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hello SamyGo, So, tls is required for wss? I thought that when I set a listen to Kamailio-Server-IPAddress:10443 and I sent a request through that IP and Port, that Kamailio would at least see the request and attempt to do something with it. I will go add tls and try again. Thank you! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Can you clarify the IP addressing scheme as you've mentioned. There is no TLS interface? Advertised address for 5060 but not for WSS interface. Both listen and advertise are public IPs ? Also,you're handling WSS requests. Do you've xlog ines in the route[xhttp:request] to view when a request lands.! My config has this: #!substdef "!MY_IP_ADDR!123.134.156.167!g" #!substdef "!MY_WS_PORT!6010!g" #!substdef "!MY_WSS_PORT!6011!g" listen=tcp:MY_IP_ADDR:MY_WS_PORT listen=tls:MY_IP_ADDR:MY_WSS_PORT Then the xhttp event route: ``` event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); #Deny any HTTP requests on any port other than WS/WSS ports. if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } #Handle HTTP(s) onwards. xlog("L_INFO", "HTTP Request Received\n"); ``` On Thu, Nov 23, 2017 at 12:18 PM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hi Sammy, First of all, thank you for taking the time to respond. Yes, port 10443 is opened. I have used this port before as asterisk’s WebRTC port and iptables shows it as open. No, I can’t even get a registration using the configuration I listed. I have an xdbg log statement right after the request_route, and I see nothing. I do know that my xdbg logs are working though because, if I register or make a call using any sip tool, I see all my logging and everything works correctly. -Steve From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:00 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hi Steve, Can you confirm that port 10443 is reachable behind the NAT to Kamailio server, validate iptables too Does your SIPml5 demo client register successfully to Kamailio? are there enough xlog lines to print out if anything lands in Kamailio. Regards, Sammy On Thu, Nov 23, 2017 at 11:34 AM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen’s in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Cannot disable EC Diffie Hellman cipher suite
by Ilyas Keskin 24 Nov '17

24 Nov '17

Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. Best regards, Ilyas Keskin

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Kamailio KEMI Framework Tutorial
by Daniel-Constantin Mierla 24 Nov '17

24 Nov '17

Hello, I started a tutorial for Kamailio KEMI Framework written in markdown format targeting use of mkdocs. It is published on github as part of kamailio-docs repository: * https://github.com/kamailio/kamailio-docs/tree/master/kamailio-kemi-framewo… The mkdocs build (to browse and read) is online at: * http://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/ It is work in progress and contributions from the community are more than welcome -- doing push requests on github project is the recommended way for contributions: * https://github.com/kamailio/kamailio-docs/ The main goal is document the role of KEMI, its benefits and how it can be used. It addition it should cover the list of functions exported to KEMI by core as well as special KEMI functions. Regarding the functions exported by kamailio modules to KEMI I added as an skeleton the ones exported by acc module. However, I am not sure if we should do it there or add them to the existing docs of the modules. Opinions on this?!? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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