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37553 discussions

Fwd: DROUTING module slow loading
by Uri Shacked 26 Oct '14

26 Oct '14

Hi, When DROUTING table "dr_gateways" is very big (40000 rows) reloading the module to memory take about 60 seconds !!! My tests show 5000 rows takes 7-11 seconds...... It is very slow. The rules table, with 40000 rows, loads immediately though. Any ideas why? Or how to get the process faster? (I know the gateway table is too big. but, this is the current implementation...) kamailio version - 3.3.6 (48db34) BR, Uri

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new compiler warning in 4.2
by Juha Heinanen 25 Oct '14

25 Oct '14

i build kamailio from latest 4.2 branch and got CC (cc) [M textops.so] textops.o textops.c: In function 'append_multibody_helper': textops.c:1867:2: warning: implicit declaration of function 'get_mixed_part_delimiter' [-Wimplicit-function-declaration] please fix asap. -- juha

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Converting JSON string to normal string
by Muhammad Shahzad 25 Oct '14

25 Oct '14

Hi, Using method json_get_field i am able to extract json string. However, this string comes with quotes which cause a hurdle in assigning to various pseudo variables, for example, -- json_get_field($redis(r=>value), "to_number", "$var(wim_to)"); json_get_field($redis(r=>value), "domain", "$var(wim_domain)"); $uac_req(ruri)="sip:" + $var(wim_to) + "@" + $var(wim_domain); -- This results in $uac_req(ruri) being set as. sip:"49XXXXXXXX"@"sip.mydomain.com" Which is obviously wrong and fails the transaction processing. Is there any way to get rid of these quotes from json strings? One possible way i can think of is using string transformation "s.substr", but there are a lot of json strings i have in config and it is too hectic to transform all of them. Is there any easier way out of this? Thank you.

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Failed to install Kamailio database
by Mahmoud Ramadan Ali 25 Oct '14

25 Oct '14

Hiii everyone, I can not create kamailio database and get this error message... any ideas ? Thanks in advance...

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How do I configure this thing?
by Nolan Darilek 24 Oct '14

24 Oct '14

OK, sorry for the basic question, but I'm overwhelmed. My install gives me a kamailio-basic.cfg, kamailio.cfg and kamailio-oob.cfg. Is there any explanation as to the difference between these? I assume "basic" is a simpler config, but I don't know what it lacks vs. kamailio.cfg, and I have no idea what "oob" means ("out-of-band?") The configs reference a kamailio-local.cfg. My assumption was that I could place config changes here, but I see later that kamailio.cfg changes some parameters. In particular, I set various auth_db parameters in kamailio-local.cfg which kamailio.cfg changes later. So I don't understand when I might want to use kamailio-local. The config files refer me to a cookbook, but unless I'm missing something, the cookbook is a long list of config directives. When I read "cookbook" I assume I'm being directed to something containing recipes like "how to enable web sockets," with step-by-step guides about which config files to edit and what to add. Am I just missing that? Is there a task-oriented getting started page that I'm missing, that would help me set up a web socket-based SIP server? Then, when I hit the inevitable issues, I can resolve specific problems rather than trying to figure this all out from scratch? I see the documentation attached to the web sockets module, but many of these docs include stand-alone config snippets with no indication as to how to integrate them into a larger config. For instance, the ephemeral auth module tells me how to set up an ephemeral auth route but doesn't give me enough context to place the snippet correctly in the 900+-line kamailio.cfg, and actually I suspect it wouldn't work anyway since I see nothing about HTTP in the kamailio.cfg. Sorry if I seem frustrated. I'm willing to learn, I've just been handed a jet engine and a few suggested tweaks, but not enough context to know how to make them. :) I'll be learning more about SIP and what role a SIP router plays, but it'd be helpful to be able to bootstrap a server so I can learn by doing as well. I'd just like to provide audio/video communication for a single SIP domain, authenticating against MongoDB and with ephemeral auth so Javascript clients can get temporary credentials. I don't want voicemail, outbound routing, etc. This seems like it should be simple to achieve, but I don't know how or where to start in a way that would allow me to build up. Thanks.

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Kamailio + RTPEngine behind NAT (STUN Handling)
by Marko Seidenglanz 24 Oct '14

24 Oct '14

Hello, We want to use Kamailio (4.2) + RTPEngine (3.3) behind NAT. Unfortunately the STUN Messages (Binding Requests) sent from remote peer (Google Chrome ) do not arrive at RTPEngine host. I think they get blocked by the firewall. RTPEngine itself does not send any STUN Messages until it get's the first message from the remote peer as Google Chrome has the "active" role in DTLS Agreement. Is there a way to tell RTPEngine to open the ports for remote peer, so that the messages arrive? When RTPEngine is not behind NAT, everything works as expected. Kind regards, Marko

2 1

Dealing with High PDD
by Ryan Brindley 24 Oct '14

24 Oct '14

Hey Community, I've tried to google and view the docs for Kamailio about fr_inv_timer (and other timers), but I don't think they do what I'm wanting. Basically, I want to cancel an INVITE request and progress to the next contact if the current contact has not send a non-100 provisional code or final response after a certain amount of time. Am I missing some obvious setting that I overlooked in my searches or do I have to approach this 'manually' (aka, not just a setting)? Any help is greatly appreciated. Ryan Brindley

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Kamailio, authentication and MongoDB
by Nolan Darilek 24 Oct '14

24 Oct '14

Hello. I'm attempting to set up Kamailio for the first time and have a few questions. 1. The auth_db module specifically calls out MySQL, Postgres and flat file databases, but makes no mention of the new MongoDB module. Is the new MongoDB module supported as well and the docs just haven't been updated to account for that? 2. In looking at the auth_db parameters, I see values for column names, but nothing for a table/collection. How do I specify what table or collection my user data is in? 3. How do I create HA1 strings? The docs state: HA1 strings are MD5 hashes of username, password, and realm. Is that: md5(username+":"+password+":"+realm) or something else? Thanks.

1 1

dispatcher behind NAT - lost ACK
by Nicholas Gill 24 Oct '14

24 Oct '14

Hello sr-users, We have Kamailio (behind NAT) configured acting as a proxy in front of some FreeSWITCH servers. There appears to be something amiss with my inbound (dispatcher) configuration which leads to misaddressing / misdelivery of the ACK after 200 OK (outbound calls appear to be proxied correctly [1]). Calls from the sip provider incoming to Kamailio are distributed by the dispatcher module to the FS server (For testing in this simplified scenario FS and Kamailio are on the same machine): SIP provider (5060) sends an invite to Kamailio(5061), Kamailio uses the dispatcher module to select a backend server and forwards the INVITE to FS(5070) (see inbound-callflow.png; kamailio.ext and kamailio.int are the same machine, just public/private addresses). I notice that at no point is a Via/record-route header for the FS server inserted into the forwarded session. I'm not actually certain this is a requirement, but I can't think of another obvious way that Kamailio could proxy the same session to the same FS server. The 100 Trying seems unremarkable [2], I suspect the 200 OK [3] is problematic. It has been proxied from the FS server, however contains no reference to the FS server address either in the via headers nor record-route (kamailio.int:5070). The incoming ACK [4] then appears to be misdelivered / lost - Kamailio receives it on the private address and forwards it to the public address rather than the FS server. My configuration [5] is built based on the default configuration + examples from the dispatcher module. There are some provisions for FreeSWITCH internal/external profiles made so the configuration listens on 2 different ports. This particular scenario should only use the 5061 port as it involves calls to an external sip provider (briefly 5060 should be proxied to FS:5080 and 5061 should be proxied to FS:5070 and vice-versa). If someone can see an issue with the configuration and/or point to an error in the call flow (i.e. should FS be inserting the Via header?) that would be greatly appreciated. Thanks, -nick [1] Outbound call flow (see also outbound-callflow.png) FS(port 5070) sends an invite to Kamailio(5061) (Kamailio is configured in FS as an outbound proxy), INVITE contains Via header for the FS server, Kamailio forwards to sip provider, and routes all messages back and forth correctly. [2] 100 Trying SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.0;rport=5060 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com> Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 INVITE Server: kamailio (4.1.6 (x86_64/linux)) Content-Length: 0 [3] Proxied 200 OK (Kamailio -> Sip Provider) SIP/2.0 200 OK Via: SIP/2.0/UDP sip.provider.com:5060;rport=5060;branch=z9hG4bKfffb.7c81ee53.0 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK Record-Route: <sip:kamailio.ext:5061;lr=on> Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr;did=0ec.84ff82b1> From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com>;tag=BFvQmggHrg74m Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 INVITE Contact: <sip:0390156842@kamailio.int:5070;transport=udp> User-Agent: IMX Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-FS-Support: update_display,send_info [4] Incoming ACK (Sip Provider -> Kamailio) ACK sip:0390156842@kamailio.ext:5061;transport=udp SIP/2.0 Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr> Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.2 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK509ZQBKB947aF Route: <sip:kamailio.ext:5061;lr=on> Max-Forwards: 69 From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com>;tag=BFvQmggHrg74m Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 ACK Contact: <sip:gw+Exetel@far.external.ip:5060;transport=udp;gw=Exetel> Proxy-Authorization: Digest username="0384171900", realm="far.external.ip", nonce="xxxx", algorithm=MD5, uri="sip:0390156842@sip.provider.com", response="xxxx" Content-Length: 0 P-hint: rr-enforced [5] Kamailio configuration (based on default configuration). Dispatcher set 1 is FS internal profile (5080) and set 2 is external profile (5070) ... #!define WITH_NAT ... alias="kamailio.int:5060" alias="kamailio.int:5061" alias="kamailio.ext:5060" alias="kamailio.ext:5061" listen=udp:kamailio.int:5060 advertise kamailio.ext:5060 listen=udp:kamailio.int:5061 advertise kamailio.ext:5061 port=5060 port=5061 ... loadmodule "dispatcher.so" modparam("dispatcher", "flags", 2) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #modparam("dispatcher", "ds_ping_interval", 60) ... modparam("nathelper", "sipping_from", "sip:pinger@etellicom.com") ... request_route { ... # Not from FS servers gets dispatched to FS servers. if(!ds_is_from_list("1") and !ds_is_from_list("2")) { route(DISPATCH); } # Requests from FS External profile routed out via specific port. if(ds_is_from_list("2")) $fs="udp:kamailio.int:5061"; ... } route[DISPATCH] { if(dst_port==5061) { $fs="udp:kamailio.int:5061"; # Hash over to uri for test purposes if(!ds_select_dst("2", "2")) { send_reply("404", "No destination"); exit; } } else { # Hash over to uri for test purposes if(!ds_select_dst("1", "2")) { send_reply("404", "No destination"); exit; } } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } ...

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sip response not handled by kamailio
by Marko Seidenglanz 24 Oct '14

24 Oct '14

Hello, We are trying to use Kamailio 4.2 with RTPEngine 3.3 behind NAT. Somehow SIP responses (200 OK) are not handled by Kamailio. The following INVITE is send to the receiving end (WebRTC Client): INVITE sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2 Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a To: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 368f6ec901a3552e3b37a70907634f4f@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Fri, 24 Oct 2014 08:38:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound ....... The client response looks like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a To: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com >;tag=E2VMB39U5NYHCXLZREFGQV2DAXPCZXNQ6J6P Contact: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com >;tag=E2VMB39U5NYHCXLZREFGQV2DAXPCZXNQ6J6P Call-ID: 368f6ec901a3552e3b37a70907634f4f@10.240.215.73:5060 CSeq: 102 INVITE Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Content-Type: application/sdp Content-Length: 938 ....... I have a xlog call as the first statement in kamailio routing script. I can see the request, which get's handled correctly but the response does not appear in kamailio log, though it arrives at the machine. I am totally helpless, what might be wrong. If we use kamailio on a public accessable machine everything works as expected. Maybe anyone might have had similiar problems using kamailio behind NAT. Kind regards, Marko

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