- sr-users - kamailio.org

call count / and work with despatcher
by jaflong jaflong 24 May '16

24 May '16

Hi Everyone, What is a accurate way to get current call count. I am only relaying calls. Initialy I tried the code below. It seemed to work, it reported the correct number of calls I had dialed. I get the result through kamcmd htable get command. ------------------------------------------ under route[RELAY] # # COUNT # if (is_method("BYE")) { $var(n) = $shtdec(callcount=>active); sht_rm_name_re("callcount=>$ft"); } -------------------------------------------- # Manage incoming replies onreply_route[MANAGE_REPLY] { # # COUNT # if (status=="200") { if (is_method("INVITE")) { # On start if($sht(callcount=>active) == $null) { $sht(callcount=>active) = 0; } # New call if ($sht(callcount=>$ft::from_tag)== $null) { $sht(callcount=>$ft::from_tag) = $ft; $var(n) = $shtinc(callcount=>active); } } xlog("L_INFO", "This is a 200 OK reply\n"); } xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } ------------------------------------------------------- However I wanted to distribute calls to several servers I further added the despatcher code below # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "-----RELAYING going to <$ru> via <$du> time [$Tf] -----\n"); This makes my counting code above give incorrect call count. How can I make call count work with dispatcher ? Do advise how I can do this Regards

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Create "Contac Groups" on the server Kamailio
by ALFONSO MARTINEZ GOMEZ 24 May '16

24 May '16

Hi, I'm working on a VoIP communication application with Kamailio as SIP server and using the SIREMIS interface. As an administrator, I can add "Subscribers" which later will be registered on the server so that they can make the necessary negotiations when they want to make a call. My question is, how you can make groups of "contacts" so that such calls can be made only with each other "contacts" in the same group? that is, we can have many "subscribers" on the server and belong to two different groups and that they can only talk to each other contact each group. Thank you so much and greetings!

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doubt using rate limit
by david escartin 24 May '16

24 May '16

Hello all i have a quick question about using module ratelimit on kamailio 4.2 or 4.4. on tests made before running it on production, i set the RED algorithm with a limit of 10 calls per second, but i saw the call limitation was not done until we got more than 20 calls per second or so on the queue. we didnt get any INVITE discarded. after surpassing around 20 INV/sec, it's true the algorithm starts to work fine and only 10INV/sec are passed more or less on production i have no easy way to check this behaviour, besides it's a feature barely used. so i only want to know if this behaviour is known, or if i'm missing soemthing my config is like this modparam("ratelimit", "timer_interval", 5) modparam("ratelimit", "queue", "4:INVITE") modparam("ratelimit", "pipe", "4:RED:10") route[RATE_LIMIT] { $var(limitation)="4"; if (!rl_check("$var(limitation)")) { send_reply("505","Limiting"); xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, rejecting with 505\n"); route(CLEAR); exit; } } thanks a lot and regards david escartin

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Please help me setting up Kamailio for android application
by Kunal Dhanda 24 May '16

24 May '16

I wanted to create my own voip application using kamailio . I am not able to find any rest integration with kamailio for my server to use for user registration and other task.Please help in developing my appplication. -- Thanks & Regards, Kunal Dhanda, Software Developer E-mail : < "kunaliitkgp09(a)gmail.com" >, Ph. No. (M) : +91 - 8884818805

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Found bug at ACC
by Pavel Eremin 24 May '16

24 May '16

Hi all, when we set syslog facility for ACC to LOCAL3, log_write_cdr() will write at LOG_DAEMON anyway. You have to recompile module acc to user another default syslog facility to save cdr with another syslog facility.

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kamctl stats shmem error changing shared memory
by Mr.Gab 23 May '16

23 May '16

Hi, i have a problem with my kamailio installation. Actually i set the shared memory to 128MB and it's all fine. When i increase the shared memory to 256Mb and more i get the following error while checking kamctl commands (like kamctl stats shmem) ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. If i revert to 128Mb kamailio is able to create the kamailio_fifo file and all works fine. Can somebody help me to solve? Thanks

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Re: [SR-Users] Change CANCEL reason when fr_timer expires
by Bruno Emer 22 May '16

22 May '16

Hello, Alex! Fisrt of all, thanks for you help with this problem! I don't think that this will help me. I can see that this parameter is enabled by default and in fact it is working in some cases. One case where this works is when the INVITE is sended for more than one endpoint (when the same user is registered in many devices). When someone answers the INVITE, Kamailio sends a CANCEL message to the other endpoints and it has the Reason header showing that the call was answered elsewhere. I want to add the CANCEL Reason header when the timer expires, so kamilio will not receive a final replay to that transaction. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-20 15:42 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a situation here where when I send a INVITE to some specific > user-agents in my environment I must change the fr_timer to a value lower > than the default. > > I can acomplish this using the "t_set_fr" commands in my cfg file, and it > works like a charm. > > The point is that when the timer reaches its limit, the kamailio server > sends a CANCEL message to the user-agent. I know that this is the normal > behavior, but I want to add the Reason field on it, so the user-agent > application can know that the call wasn't answered on time. > > I was looking the TM module documentation to understand how to do this, > but I couldn't figure out a way to get this working. > > Is there a way to add the Reason in CANCEL messages generated by the > fr_timer? > > > Thanks. > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >

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Change CANCEL reason when fr_timer expires
by Bruno Emer 21 May '16

21 May '16

Hello all! I have a situation here where when I send a INVITE to some specific user-agents in my environment I must change the fr_timer to a value lower than the default. I can acomplish this using the "t_set_fr" commands in my cfg file, and it works like a charm. The point is that when the timer reaches its limit, the kamailio server sends a CANCEL message to the user-agent. I know that this is the normal behavior, but I want to add the Reason field on it, so the user-agent application can know that the call wasn't answered on time. I was looking the TM module documentation to understand how to do this, but I couldn't figure out a way to get this working. Is there a way to add the Reason in CANCEL messages generated by the fr_timer? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

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extra characters in re-invite "TO" field
by Al S 20 May '16

20 May '16

Hi, Kamailio adds extra characters in the re-invite "TO" field during session timer. There is no header manipulation in our route[WITHINDLG] block. Re-INVITE received by KamailioTo: <sip:8131111111@?.?.?.?;otg=1>;tag=1c1048632956 Re-INVITE forwarded by KamailioTo: <sip:8132222222@?.?.?.?;T\033\023Z\f1>;tag=1c1048632956 Your help would be greatly appreciated. Thanks,AS

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Question about testing RTPENGINE
by Dmitry 20 May '16

20 May '16

Hello I am trying to make a test setup to test the RTPENGINE. Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.132)-----via NAT-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) First WITHOUT RTPENGINE : When A calls to B - the sound both ways, so it is ok. Second WITH RTPENGINE: When A calls to B - the sound both ways, so it is ok. Actually what I want is to make such setup that calls without the RTPENGINE are impossible.

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WebRTC
by Moacir Ferreira 20 May '16

20 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

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Re: [SR-Users] Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 19 May '16

19 May '16

Is there is reason why this post does not get approved? From: Pejman Rajabian [mailto:pejman@italkip.com] Sent: Wednesday, May 18, 2016 12:59 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio notify conflict when a second call lands on a phone I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 pejman(a)italkip.com

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Sipcapture on SPARC
by Spencer Thomason 18 May '16

18 May '16

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some sore of endianness issue? Thanks for any assistance! Regards, Spencer

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[kamailio 5.0] upgrade VS rollback/backward compatibility
by Alex Lutay 18 May '16

18 May '16

Hi, We are heavily using Kamailio in our product Sipwise NGCP and currently testing possibility to provide rollback for the customers with the huge systems as the last resort functionality in case if upgrade went wrong. We hope never use it on practice but prefer to have such possibility. The main logic here is to change DB in 'backward compatible' way. So OLD code must work with NEW DB schema (but NOT a vice verse). We have some rules which helps us care about it, like "Do not drop columns, just discontinue them" and "All new fields should have defaults", etc. Also we have separated 'code', 'configs' and 'data' parts of our product and rollback 'code'/'configs' only, while 'data' (e.g. MySQL DB) stays in new state. Otherwise rollback will require a lot if time to revert all DB statements and sometimes it is just not possible. While every time we have to upgrade Kamailio we have a problems with rollback possibility. The point here that Kamailio checks DB table_version and doesn't start if the version mismatch found. Q: What about to add Kamailio possibility to work in backward compatible mode? Sample for upgrade Kamailio 4.3.x -> 4.4.x (the versions here as an example only!) https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0 We have 3 DB statements there which are adding new columns to 3 tables, like > -- table: active_watchers > ALTER TABLE active_watchers > ADD COLUMN flags INT(11) NOT NULL DEFAULT '0', > ADD COLUMN user_agent VARCHAR(255) NOT NULL; > DELETE FROM version WHERE TABLE_NAME='active_watchers'; > INSERT INTO version (`table_name`, `table_version`) VALUES ('active_watchers','12'); Problem here is missed default for column 'user_agent' which can be empty value as for me. So, lets assume we have it there. Then we can add new column to table 'version', like that: > mysql> select * from version where table_name='active_watchers'; > +-----------------+---------------+---------------------+ > | table_name | table_version | table_compatibility | > +-----------------+---------------+---------------------+ > | active_watchers | 12 | 11 | > +-----------------+---------------+---------------------+ > 1 row in set (0.00 sec) Currently: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 Proposed behaviour: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 and compatibility option is not set - if compatibility option is set, check value of table_compatibility and refuse to load if current version less then table_compatibility value >From the sample above I would say module presence in Kamailio 4.4.x is compatible with Kamailio 4.3.x if we define default value for the field 'user_agent'. Having that will allow us to rollback on previous NGCP release and _temporary_ work on Kamailio 4.3.x having new Kamailio DB 4.4.x P.S. This is just a question to check Kamailio community vision here and reality is more complex as described, so some additional checks/limitations will be in place. P.P.S Compatibility Kamailio option can/should be disabled by default. Field 'table_compatibility' can be outside table 'version', e.g. in separate table. Complete compatibility logic can be some separate Kamailio 'compatibility' module. Thank you! -- Alexander Lutay Head of Quality Assurance Sipwise GmbH, Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Office: +43(0)13012036 Email: alutay(a)sipwise.com Website: https://www.sipwise.com Meet us @ ANGACOM: Hall 10.1 / booth N10 Exhibition and Congress for Broadband, Cable & Satellite: 07 – 09 June, 2016 in Cologne

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Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 18 May '16

18 May '16

I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 <mailto:pejman@italkip.com> pejman(a)italkip.com

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invalid route error when turning on Roles in kamailio kazoo
by Dmitry 18 May '16

18 May '16

hello When I uncomment in /etc/kazoo/kamailio/local.cfg#!trydef TRAFFIC-FILTER-ROLE (it was commented previously) And the kamailio does not start and the log gives this error:-Invalid route satement The structure of files is simple:the trafic-filter-role is in the separate file. and in the main file (default.cfg #!ifdef TRAFFIC_FILTER-ROLEinclude_file "traffic-filter-role.cfg" and that's it. But it gives INVALID ROUTE ERROR

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 17 May '16

17 May '16

Hello, David. Sorry for the time to reply you. I was a little busy here in the last few days. I am not sure about the "==". We had tried it here, but it only worked when we used with =~. We are currently running kamailio 4.2, so it may be a problem with our version. About the $hdr variable, it is used to get information from the SIP Headers. I am not sure, but I dont think that there is an header with Instance-ID, so you must try to find how can you get this ID in kamailio, as I don't know where is it saved. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-13 12:24 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > After some time I was able to get this working here, and now I am coming > back to share my configuration with you guys (in case someone needs to get > this working). > > Here is what we have done: > > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > > xlog("L_INFO", "checking $tu registered with > User-Agent:$hdr(User-Agent)\n"); > if(reg_fetch_contacts("location", "$tu", "caller")) > { > xlog("L_INFO", "$(ulc(caller=>count)) records > found for $tu\n"); > $var(i) = 0; > while($var(i) < $(ulc(caller=>count))) > { > > xlog("L_INFO", "record $var(i) with > ruid:$(ulc(caller=>ruid)[$var(i)]) - > User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); > if > ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) > { > $var(ruid) = > $(ulc(caller=>ruid)[$var(i)]); > xlog("L_INFO", "unregistering $tu > ruid:$var(ruid)"); > unregister("location", "$tu", > "$var(ruid)"); > } > $var(i) = $var(i) + 1; > } > } > if (!save("location", "0x00")) > sl_reply_error(); > > exit; > } > } > > > There are some logs over there because we used it to see what was going on > while testing the configurations. > > > With this we can get just one register per User-Agent. > > > And again, thanks Olle for your help in this problem > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Sure it is! >> >> This is exactly what I was looking for! Actually I wasn't able to see >> that there was a way to perform this action! >> >> Thanks for your help, Olle! >> >> I will try to create this, and when I have it done, I will post my script >> here, so if someone else need the same problem I can help. >> >> Now, I have just one more question about my scenario: when I register a >> user, is there a way to create something like a "custom field" on the >> location? To explain better: If possible, I can add something like a custom >> header that says if the user is registered from a phone device or the web >> interface, then I can perform the loop and replace only the oldest contact >> using that device... >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> This is my problem... I want to have 2 contacts for AOR, and not only >>> one. >>> >>> When the 3rd arrives, it must remove the oldest and continue with 2... >>> >>> *If I am using the 0x04 parameter, it will save just one contact, and >>> the user will not be able to get the Invite both, in the mobile device and >>> web interface.* >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >>> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello, *Olle!* >>>> >>>> *Thanks for your help in this case, and don't worry about the time >>>> (actually I wasn't even expecting to get an answer today).* >>>> >>>> >>>> *I know the max_contacts parameter. But the problem is that if I set >>>> the max_contacts to 2, when the user tries to register again, he gets an >>>> error message 503. The point is that I want to get the user registered, and >>>> replace the oldest location entry for the newest one.* >>>> >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>>> >>>>> Hello all. >>>>> >>>>> I have problem here and I tried to find a solution and search over >>>>> internet, but without success. >>>>> >>>>> My scenario is the following: I have an application that must be >>>>> registered in Kamailio when a user logs in the web interface, so he can get >>>>> calls (something like a web softphone using webrtc). At this point, we are >>>>> OK, and everything is working fine. >>>>> >>>>> To get these register functions I am using the parameter >>>>> "save("location", "0x04")" as described in the REGISTRAR module >>>>> documentation, so if a user logs in another web browser or computer, only >>>>> the last one will continue registered and all calls will be forwarded to >>>>> him. >>>>> >>>>> The point is that now we are creating a phone app that will do almost >>>>> the same thing as the web interface, allowing users to receive calls using >>>>> the mobile device, and here is my problem: I want to allow my users to be >>>>> registered on two devices at the same time, but if a user logs into another >>>>> device, I don't want to reply with a 503. I want to allow the user to >>>>> register again, deregistering the oldest contact. >>>>> >>>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>>> description it says "implemented overwrite oldest contact behaviour", >>>>> but I couldn't find any documentation about it. >>>>> >>>>> So, is there a way to get this working today? >>>>> >>>>> >>>>> *Bruno Emer* >>>>> >>>>> Mobile: +55 11 96540-0044 >>>>> email: brunoemer(a)gmail.com >>>>> >>>> >>>> >>> >> >

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Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Enum module and parallel branching
by Loic Chabert 17 May '16

17 May '16

Hello, I'm trying to use enum module at core level to route my numbers to end-point asterisk servers. As i read (and validated by tests), when a phone number has two DNS NAPTER entries, enum module create two branch, one for each entries, and send request to asterisk servers (in the same time). Do you know any tricks or functions to disable parallel branching, and make serial branching instead (following responses returned by asterisk) ? I have also change preference and order in dns config file, same results. Thanks in advance. Regards.

2 3

MySQL database for kamailio as S-CSCF
by Christoph Valentin 17 May '16

17 May '16

Hi Community, I'm doing very first steps to use kamailio as an S-CSCF. Now I'm confused about some defines in the examples/scscf/scscf.cfg file. #!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" Tutorials/ims/Installation-howto tell me only I-CSCF needs a database, I do not find anything about database for S-CSCF. However, when I comment out the defines ##!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" then the kamctl start results in a segmentation fault. I have installed kamailio-4.4.0 from RPMs. Is there any hint about database for S-CSCF? Thanks in advance Christoph

1 1

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by M R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Debian, dnssec.so, and missing dnsval.conf
by David Holl 17 May '16

17 May '16

In the interest of sharing and promoting DNSSEC, I was recently stumped on a problem where I noticed I could not call sip:test.echo@sip5060.net without disabling the dnssec.so module in my kamailio.cfg. The last time I had tried the sip5060 echo test was several months ago, and in the intervening time, I had applied the numerous debian package updates. Thus, I could not isolate the cause of the problem to one of the many package updates such as kamailio 4.4.x, libval, my own DNSSEC dns server, etc... My initial only clue was kamailio's syslog messages: May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve "sip5060.net" :unresolvable A or AAAA request (-7) May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) To summarize the problem, I found that kamailio's dnssec.so was failing queries to domains that were not DNSSEC protected such as sip5060.net. This behavior is a bug, since sip5060.net is in this case, provably not participating in DNSSEC from the toplevel "net." zone [1]. The fix was to provide a default configuration file to libval, which was somehow not included in Debian's libval15 package. I created the following configuration from the default dnsval.conf in the source distribution: https://dnssec-tools.org/download/dnsval-2.2.tar.gz https://dnssec-tools.org/download/dnsval-2.2.tar.gz.asc as well as their wiki: https://www.dnssec-tools.org/wiki/index.php/Dnsval.conf cat << 'EOF' > /etc/dnssec-tools/dnsval.conf ################################## # Global Options ################################## global-options trust-oob-answers yes # I'm running a DNSSEC nameserver locally, # so 4 KiB queries to localhost should be OK: edns0-size 4096 env-policy disable app-policy disable rec-fallback no ; : trust-anchor . DS 19036 8 2 49AAC11D7B6F6446702E54A1607371607A1A41855200FD2CE1CDDE32F24E8FB5 ; : zone-security-expectation . validate ; : provably-insecure-status . trusted ; : clock-skew . 0 ; EOF In summary: I find it odd that a default dnsval.conf was not included with my system's normal installation of the "libval15:amd64" package, version 2.2-4. After providing dnsval.conf, kamailio's dnssec.so was happy again. Side notes: * To identify the directory for dnsval.conf, run libval-config --dnsval-config On Debian, libval-config is in the libval-dev pacakge, and this package includes a man page for dnsval.conf as well. * To test your libval and dnsval.conf, try dt-validate -s On Debian, dt-validate is in the libval-bin package. - David [1] -- Authenticated denial of existence: https://tools.ietf.org/html/rfc5155 https://tools.ietf.org/html/rfc7129 And for sip5060.net, there's an NSEC3 record in the top level net. zone proving non-existance of sip5060.net DS records: http://dnsviz.net/d/sip5060.net/dnssec/ Translation: sip5060.net provably not enabled DNSEC.

2 1

kamailio not authenticate a subscribed username
by Al S 16 May '16

16 May '16

Hi,1) A generic subscriber is added to kamailio database:kamctl add pbx password2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call:UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it)The correct password is being sent to kamailio.Your assistance is greatly appreciatedThanks,AR

1 0

kamailio not authenticate a call with subscribed username
by Al S 16 May '16

16 May '16

Hi, 1) A generic subscriber is added to kamailio database:kamctl add pbx password 2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call: UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it) The correct password is being sent to kamailio. Your assistance is greatly appreciated Thanks,AR

1 0

Re: [SR-Users] Another question about RTPengine module
by Dmitry 16 May '16

16 May '16

the problem is: rtpengine was compiled with USE_SCTP,but kamailio that was installed via RPM is without this flag. How to tell compiler NOT TO " USE USE_SCTP" flag? +++++++++++++++++++++++ Makefile.defs:SCTP ?= 0 - and it compiles without sctp. Now kamailio.log:ERROR: rtpengine [rtpengine.c:1113]: child_init(): Name or service not knownERROR: <core> [sr_module.c:929]: init_mod_child(): Error while initializing module rtpengine (/usr/lib64/kamailio/modules/rtpengine.so) View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users On Sunday, May 15, 2016 6:21 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: yes, I wrote that modparam are recognized.MODPARAM must be placed i the sae place where the RTPPROXY resides in the config.+++++++++++++++++++++++++++++++ Another question:kazoo kamailio KAMAILIO-V :4.3.5:kamailio -vversion: kamailio 4.3.5 (x86_64/linux)flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 To add the RTPENGINE - I found Kamailio 4.3.5 sources https://www.kamailio.org/pub/kamailio/4.3.5/src/ When I start kamailio I get the following error:ERROR: <core> [sr_module.c:431]: version_control(): module compile flags mismatch for /usr/lib64/kamailio/modules/rtpengine.so I did the following to add missing modules: #make include_modules="rtpengine topoh" cfg #make all cp /usr/src/kamailio-4.3.5/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so How to fix it? The sources version seems to be the same On Saturday, May 14, 2016 1:21 AM, ycaner <yasin.caner(a)netgsm.com.tr> wrote: Hello; i am not sure. i downloaded from git kamailio. add rtpengine instead of rtpproxy . it works fine. it is hard to solve problem without log. Cheers. try to run with kamailio -E -dd , maybe it gives information. check kamailio.cfg is that rtpengine is loaded like loadmodule "rtpengine.so" 2016-05-13 11:35 tarihinde Dmitry [via SIP Router] yazdı: one more question about rtpengine. I installed Kamailio (centos 6.7) from source as You recommended make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg no rtpengine daemon (it is not installed) but the picture is the same: modparam ("rtpengine,....) kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(19523) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found it means that it does not depend upon rtpengine daemon. # single rtproxy modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") I copied it from http://www.kamailio.net/docs/modules/4.2.x/modules/rtpengine.html STRACE kamctl start -says nothing anout rtpengine What else is it possible to do to find the cause of the problem? On Friday, May 13, 2016 11:45 AM, ycaner <[hidden email]> wrote: Hello; rtpengine module isnt default. you need to add with "make include_modules". after make && make install , automatically it will be copied to lib/modules. Cheers. 2016-05-13 09:42 tarihinde Dmitry [via SIP Router] yazdı: And one more question on it: If the Kamailio was installed from RPM and rtpengine.so is missing - May I add rtpengine.so the following way: make include_modules="rtpengine" cfg make all and then copy rtpengine.so to /modules/ directory Is this way (adding missing modules) is correct? On Friday, May 13, 2016 11:25 AM, Dmitry <[hidden email]> wrote: hello I tested on Centos 6.7 as well as Ubuntu server - the problem is the same. RTPENGINE was installed both from dpkg and from Source (manual compilation) So I need a clarification on it. RTPENGINE daemon is started. The Kamailio module itself is taken from /usr/src/kamailio, where the source was compiled. mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so sudo /etc/init.d/kamailio restart | grep rtp ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <rtpengine> found version: kamailio 5.0.0-dev4 (x86_64/linux) Any help is appreciated. On Thursday, May 12, 2016 6:04 PM, Dmitry <[hidden email]> wrote: Hi, I installed Kamailio from GIT. I did make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg Actually loadmodule "rtpengine.so" does not affect kamailio start/restart. modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") causes this failure: [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found On Thursday, May 12, 2016 12:26 PM, ycaner <[hidden email]> wrote: Hello; on bash , kamailio -V gives Version , kamailio -c gives debug result. you can ask from here https://groups.google.com/forum/#!forum/2600hz-dev for kazoo rpm. if there is something wrong , they will fix it. you can install from source code from here => https://www.kamailio.org/w/ and "make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg " in kamailio folder before 'make all' , helps you add modules. if there is a dependecies is missing , gives error. Cheers. 2016-05-11 19:51 tarihinde Dmitry [via SIP Router] yazdı: I installed Kamailio : yum install -y kazoo-kamailio https://2600hz.atlassian.net/wiki/display/Dedicated/via+RPM git clone https://github.com/sipwise/rtpengine.git the latest version of rtpengine and I do not know how to view the version of Kazoo-Kamailio On Wednesday, May 11, 2016 6:31 PM, ycaner <[hidden email]> wrote: hello ; maybe some devel are missing. which version using?i will try. cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML --

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