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37543 discussions

How to use the ip parameter in rtpproxy_manage()
by Sebastian Damm 29 Aug '14

29 Aug '14

Hi, I have the following setup: UAC ---> LB ---> Proxy ---> GW In NAT scenarios the loadbalancer detects it, but the proxy communicates with the RTP proxy. I want to send the original caller IP as detected by the loadbalancer (and transported to the proxy) to the RTP proxy. As far as I understand the documentation of the rtpproxy module, I could call rtpproxy_manage with a second parameter, indicating which IP address should be sent to the RTP proxy. I tried sending a pseudovariable, both within quotes or not, or even a static string. But when I look at those messages sent to the RTP proxy, there's always the IP from where the SIP packet was received (the loadbalancer IP) in the request. Previously we used the "r" parameter, sending the original IP from the SDP to the proxy. But due to strange behavior of some UACs, we want to get rid of that. Do I misunderstand the second parameter? This is what the documentation says: * ip_address* - new SDP IP address. Unfortunately, I couldn't find any example using this. Best Regards, Sebastian

2 3

Help analysing segmentation fault
by Charles Chance 29 Aug '14

29 Aug '14

Hi All, I wonder if some one could help me to diagnose a recurring issue? It happens at random times/intervals and under varying load. But always, just before the time of crash, I see the same critical error in log: CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0xb0632134 [1367:5814] with clid ' 0695dd7a346188dd24e7520e6c01092c(a)sip.sipcentric.com' and tags 'as77c89620' '' Analysing the core dump reveals: Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 128 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 0x081a737c in parse_uri (buf=0x3a70006e <Address 0x3a70006e out of bounds>, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389 389 scheme=buf[0]+(buf[1]<<8)+(buf[2]<<16)+(buf[3]<<24); (gdb) frame 1 #1 0x008fe5dd in dialog_publish (state=0x903f37 "Trying", ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275 275 if (parse_uri(ruri->s, ruri->len, &ruri_uri) < 0) { (gdb) p *ruri $1 = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275} (gdb) up #2 0x008ff277 in dialog_publish_multi (state=0x903f37 "Trying", ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:387 387 dialog_publish(state,&(ruris->s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck); (gdb) p *ruris $2 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} (gdb) up #3 0x0090187a in __dialog_created (dlg=0xb0632134, type=2, _params=0x6db064) at pua_dialoginfo.c:470 470 dialog_publish_multi("Trying", dlginfo->pubruris_caller, &(dlg->from_uri), (include_req_uri)?&(dlg->req_uri):&(dlg->to_uri), &(dlg->callid), 1, dlginfo->lifetime, 0, 0, 0, 0, send_publish_flag==-1?1:0); (gdb) p *dlginfo->pubruris_caller $3 = {s = {s = 0x31590014 <Address 0x31590014 out of bounds>, len = 275}, next = 0xb0b5fd00} (gdb) p *dlginfo->pubruris_caller->next $4 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} In config, for pua_dialoginfo we are enabling the option "use_pubruri_avps" and setting "pubruri_caller_avp" and "pubruri_callee_avp" accordingly. Therefore, in pua_dialoginfo.c it is using get_str_list() function to set dlginfo->pubruris_caller from the avp. Could this be some race condition or something completely different? Thanks in advance, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

3 7

Add websocket header to SIP
by Kamrul Khan 28 Aug '14

28 Aug '14

Hi, I want to append the origin header of my incoming websockets to the SIP INVITES. I tried something like the below but didnt work: if (ws_handle_handshake()) { if(method=="INVITE"){ insert_hf("Origin: $hdr(Origin)"); } exit; } Is there any other way to do it ?

2 3

[Kamailio-4.1.5] [/etc/init.d/kamailio/start] Error
by Djamel Bahamid 28 Aug '14

28 Aug '14

Hello, When I try to execute the /etc/init.d/kamailio start, I've this error : [root@gtk:~][Mon 18/08/2014 16:21:58] # /etc/init.d/kamailio start Not starting Kamailio SIP server: invalid configuration file! failed failed ERROR: loading config file(/usr/local/etc/kamaiilio/kamailio.cfg): No such file or directory failed failed Thank you for your help, Best Regards, Djamel BAHAMID.

3 5

rfc: distributing dialog profiles
by Daniel-Constantin Mierla 28 Aug '14

28 Aug '14

Hello, if you are on sr-dev list, you should have seen the last commits to dialog module, which add support for remote profiles. As most of you know (or at least the users of dialog module), profiles are used to group active calls, their main purposes being counting dialogs belonging to a profile for implementing active call limits (e.g., all calls in the system, call from/to a specif user/address). It was working on a single Kamailio instance, but one could use other meanings to interact with other kamailio instance to get the number of calls across multiple instances. The purpose of remote profiles is to make the whole process easier and more transparent for config file, simplifying the maintenance. A remote profile is an item stored in local profile list not bound to a local dialog. At this moment, dialog module in master branch exports a function dlg_remote_profile() that allows to add/remove a remote profile. The next step is to build the notification system between kamailio instances. There are couple of options, as well as questions, that I want to discuss before going to implement the mechanism. There are two major aspects to care of: A) Distribution, I thought of two options for it: 1) using dmq module to publish the operations done with local profiles (add/remove) -- this seems to be the natural choice right now, Charles Chance is also willing to put effort in this direction This will be coded inside the module, so config won't be affected much (eventually some extra parameters of dialog flags). 2) add event routes for adding/removing a local profile. The attributes of the profile can be made available via some PV, then in config use what so ever option (e.g., http query, json rpc, etc) for distributing the information. This will require changes in config file and might not be trivial for new users B) Persistence, with the options: 1) store remote profiles in local database. It is good for restart, but if there is big downtime, then some notifications can be lost and the list of remote profiles might be inaccurate 2) keep in memory only and at startup send a 'request-for-update' to the other instances -- these instances will send the list of their local profiles. It can take a bit between request-for-update and receiving data from the other servers -- during this interval, the checks of overall call limits might not give the desired answer Of course, there can be enhancements/combinations, but it has to be analysed if the benefit worth the complexity. For 1), store also the IP of each remote profile and then interrogate for each profile to see if its still active. For 2), add a mechanism to reject new calls until the request-for-update is completed. Anyone having other ideas or comments to the above? Note that each profile got a unique id, so if you upgrade your dialog module, profiles for existing active calls will not be loaded. Also, the code was barely tested, report here any issue you have with dialog from master branch. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

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Calls to devices behind NAT not working
by Adnan 28 Aug '14

28 Aug '14

Hi all, I'm trying to configure a Kamailio server, and am having trouble getting devices behind NAT receive calls. Looking at a SIP trace, nothing is being received by the end point, even though Kamailio seems to be sending out a response. I've opened up my firewall rules, so that shouldn't be an issue. If I call another endpoint which is behind the same NAT, it works (although no audio), and looking at the trace, the SIP messaging is coming from the Kamailio server (ie. not the other endpoint). I've even tried using the default kamailio-advanced.cfg & kamailio-basic.cfg both of which didn't work either. I've attached the current config I'm working on, if someone could please have a look through it, and see where I'm going wrong I'd highly appreciate it. Thanks Adnan

1 0

Minimal configuration without database support
by Eugene Prokopiev 27 Aug '14

27 Aug '14

Hi, Is it possible to run kamailio without database support (even dbtext)? Where can I find minimal configuration example with minimal modules set only for forwarding any requests/responses from one ip address to another one? -- WBR, Eugene Prokopiev

5 23

Understanding hashtable usage
by aft 27 Aug '14

27 Aug '14

Hi, I've got a piece of config code which should solve my problem, but i'm not getting the code. It involves hashtable module. Here it goes : modparam("htable", "htable", "a=>size=8;autoexpire=400") modparam("htable", "htable", "b=>size=8;autoexpire=31") Ok we have two hashtable , a and b, both has a "size" of 8. Then, else if(status == "503") { if($sht(a=>alarm::503) == $null) $sht(a=>alarm::503)=0; $sht(a=>alarm::$si::503) = $sht(a=>alarm::$si::503) + 1; } Here i get lost. what's "a=>alarm::503" and what's "a=>alarm::$si::503"? What's there difference? Next the code goes like :(its supposed be pseduocode) $elem = the size of the sht array. while($i < $elem) { $src_ip = $sht(source_ip_array)[$i]; $i = $i +1; sql_query("cb", "INSERT INTO alarm_data (create_date, type, total, description, source_ip) VALUES(NOW(), 'Too Many 503', $sht(a=>alarm::$src_ip::503), 'service unavailable', '$src_ip')"); } Now i have couple of problems with this code [1] what should be the value of $elem? [2] $sht(source_ip_array)[$i]? what should be the value of source_ip_array? The thing i'm trying to do is, when a 503 (Service unavailable) comes to kamailio, I want to insert it into a database which records this. -- -Cheers -Arif

1 0

match_dialog not handling BYE
by Satish Patel 26 Aug '14

26 Aug '14

I really want to use top hide using dialog module but having very difficult time.. [UA]------[Opensips]-----[Asterisk] I want to hide UA so Asterisk doesn't know its IP. I have following config, Please correct me if anything wrong in my config, Problem is if callee whoever connected to asterisk hangup call it doesn't not know where to send BYE because of top hiding remove all return routes.. i am surprise how it is working with other folks? something must be wrong in my config then.. if (loose_route() || match_dialog()) { if ($DLG_status==NULL) { xlog(" cannot match request to a dialog \n"); # something wrong - might want to drop such requests } if (is_method("BYE") { setflag(ACC_DO); # do accounting ... setflag(ACC_FAILED); # ... even if the transaction fails acc_aaa_request("Internally generated BYE"); end_media_session(); } else if (is_method("INVITE")) { # even if in most of the cases is useless, do RR for # re-INVITEs alos, as some buggy clients do change route set # during the dialog. record_route(); } .... .... if ( uri=~"^sip:[0-9]*@.*") { uac_replace_from("sip:4545@65.xxx.xxx.xxx"); t_on_failure("3"); resetflag(7); topology_hiding(); t_relay( "udp:65.xxx.xxx.xxx" );

1 0

Not able to start tls on port 5061
by Debojit Kakoti 26 Aug '14

26 Aug '14

I am getting error INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... -- Debojit Kakoti Mobile : +91 9650654870 Email : debojkakoti88(a)gmail.com Skype : deboj1988

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