sr-users August 2014

sr-users@lists.kamailio.org

73 participants
131 discussions

Presence - How to force Kamailio to use one subscription per user?
by Veerabhara Gundu 30 Aug '14

30 Aug '14

Hello Friends, I have a scenario, where clients are going away and re-registering with new IP and Port numbers. Presence server keep sending NOTIFY messages for both old Subscription and new Subscription for same clients till the old subscription expires. It's unnecessary traffic and processing on both client and server. I am looking for a way to force Kamailio to use one subscription per user similar to mode:04 of location table. If it's not possible, looking for a way to remove subscription from config file when I detect this error.We have an option to remove automatically Subscription when NOTIFY failed, but this is not working for TCP based clients. In TCP mode, when Kamailio not event trying to send NOTIFY if Kamailio already knows that TCP connection is broken and replying with 500 error. Any alternative suggestion also welcome. Thanks, Veera Thanks, VG

1 3

What does "::" operator do in statements related to hashtable?
by aft 30 Aug '14

30 Aug '14

Hi, >From kamailio documentation, the usage of hashtable is given as : modparam("htable", "htable", "a=>size=4;") ... $sht(a=>test) = 1; $sht(a=>$ci::srcip) = $si; I get it that in the first statement, a is the hashtable, a new key-value pair is added to one of its empty bucket (test,1). What is happening in the second statement? -- -Cheers -Arif

3 3

Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS
by Alex Villacís Lasso 29 Aug '14

29 Aug '14

As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration. In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost): REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad>";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0 Asterisk answers this through UDP, and Kamailio forwards it through the websocket: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0 Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket: NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Since I have not implemented handling of voicemail indications, the browser answers this: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;tag=ggu5etber9 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0 After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws . I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?

2 5

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.

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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

5 9

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

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sr-users August 2014