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sr-users@lists.kamailio.org

42 participants
37519 discussions

newbie INVITE routing logic help
by Skyler 24 May '11

24 May '11

Hi all, Need some newbie advice on how to write INVITE routing for my scenario. I currently have kamailio+asterisk-realtime integration working in the following way: Outbound calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> SIP/PSTN This works, though I'd prefer to use Kamailio + LCR and gateway failover as Asterisk doesn't do this well. Here's what I want to do: (for now, small steps) Outbound Calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> Kamailio(3.1.3)+(lcr) -> SIP/PSTN I know I have to set kamailio as a GW in asterisk instead of the PSTN GW's and add the PSTN GW's to kamailio instead, though I'm at a loss as to how to write the routing logic so that kamailio sends calls from asterisk to PSTN. route[LOCATION] { ...non-INVITE check ends here... if(route(FROMASTERISK)) { # coming from Asterisk --- Does this mean check kamailio local users first before continuing? if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } ## -- If match 11 digits and starts with '1' or if match 10 digits and starts with [2-9] if($rU=~"^(1\d{10})$") || if($rU=~"^([2-9]\d{9})$") { xlog("L_INFO", " FROMASTERISK: This will become route(NORTHAMERICA); \n"); exit; ## to be removed } else { xlog("L_INFO", " FROMASTERISK: This will become route(INTERNATIONAL); \n"); exit; ## to be removed } # new call - send to Asterisk route(TOASTERISK); } Will this logic work? I haven't determined how to write route(NORTHAMERICA) or route(INTERNATIONAL) yet so I'm just logging to debug the logic and syntax. Thanks in advance.

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Kamailio to handle call states
by Yeris Antonio Madrigal Castro 23 May '11

23 May '11

Hi I am planing to change my old T1-E1 pbx, wich is AVAYA, for an asterisk cluster. I am planning this for a call center and I spec 80 currents calls for 30 agents sometimes, so I need to handle big queues and call and agent states, is it possible for kamailio handle the call states and agent states, and leave asterisk for the call center functions? Thanks in advance

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TLS - Kamailio Asterisk - Registration issue
by Lucas Alvarez 23 May '11

23 May '11

Hi, I have confgured Kamailio with TLS in an Kamailio-Asterisk implementation. The phones (Bria - Eyebeam - Aastra 57i) are registering in Kamailio without problems, but the registration is not being forwarded to Asterisk. Kamailio is listening on port tls:5061 only and Asterisk on udp:5080. Should Kamailio has to listen on port 5060 also for UAC module functions? If I enable the port 5060, how do I prevent common SIP - UDP registrations? I mean, I want the registrations to be over TLS only. I would appreciate if someone can put me on the right path. Thanks in advance. Lucas Alvarez

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change ptime problem
by hala alramli 22 May '11

22 May '11

i am trying to route to pstn provider,but i faced a problem . my provider accept ptime 20 while i am send the call with ptime 30 how i can change ptime i have kamailio with freeswitch any help please

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Kamailio + Asterisk 1.6.2 Realtime - REGFWD : 401 not authorized
by Skyler 22 May '11

22 May '11

Hi all, After 3 hours stuck on this I have to ask the group. I am setting up Kam 3.1.3 +Ast 1.6.2.18 + realtime following Daniel's guide on the Asipto site. The problem I see is 401 not authorized when uac tries to register. Below is ngrep . do you see something I am missing or have any pointers? I can't seem to get any devices to register with asterisk. --- TIA interface: any filter: (ip or ip6) and ( port 5060 ) U 2011/05/21 23:40:39.333181 192.168.1.132:5060 -> 192.168.1.104:5060 REGISTER sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-cd60fa71. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: testing <sip:4444@192.168.1.104>. Call-ID: 5407498c-e755970e(a)192.168.1.132. CSeq: 55278 REGISTER. Max-Forwards: 70. Authorization: Digest username="4444",realm="192.168.1.104",nonce="Tdiwvk3Yr5JAcX9Iljh/AX0BieRt73Y v",uri="sip:192.168.1.104",algorithm=MD5,response="68bc7ad2de02caa48b8cbb22b 6de7aa6". Contact: testing <sip:4444@192.168.1.132:5060>;expires=30. User-Agent: Linksys/PAP2-3.1.23(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. U 2011/05/21 23:40:39.347248 192.168.1.104:5060 -> 192.168.1.132:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-cd60fa71;rport=5060. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: testing <sip:4444@192.168.1.104>;tag=d0c53e97c845e2d86a9876bb195e3450.e92e. Call-ID: 5407498c-e755970e(a)192.168.1.132. CSeq: 55278 REGISTER. Contact: <sip:4444@192.168.1.132:5060>;expires=60;received="sip:192.168.1.132:5060". Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. U 2011/05/21 23:40:39.349726 192.168.1.104:5060 -> 192.168.1.104:5080 REGISTER sip:192.168.1.104:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK0f62.635b95f3.0. To: sip:4444@192.168.1.104. From: sip:4444@192.168.1.104;tag=a707bfce77c9367d1734afac7b3de6ca-f4de. CSeq: 10 REGISTER. Call-ID: 7f1ddb01-6717(a)192.168.1.104. Content-Length: 0. User-Agent: kamailio (3.1.3 (i386/linux)). Contact: <sip:4444@192.168.1.104:5060>. Expires: 30. U 2011/05/21 23:40:39.351790 192.168.1.104:5080 -> 192.168.1.104:5060 SIP/2.0 401 Unauthorized. v: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK0f62.635b95f3.0;received=192.168.1.104. f: sip:4444@192.168.1.104;tag=a707bfce77c9367d1734afac7b3de6ca-f4de. t: sip:4444@192.168.1.104;tag=as64ca9fe3. i: 7f1ddb01-6717(a)192.168.1.104. CSeq: 10 REGISTER. Server: taridium ipbx. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. k: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.104", nonce="5bad77f5". l: 0. U 2011/05/21 23:40:53.322348 192.168.1.104:5060 -> 192.168.1.132:5060 .... U 2011/05/21 23:40:57.180694 192.168.1.132:5060 -> 192.168.1.104:5060 NOTIFY sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-56c97474. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: <sip:192.168.1.104>. Call-ID: dbc8d986-9fbb95e(a)192.168.1.132. CSeq: 47 NOTIFY. Max-Forwards: 70. Event: keep-alive. User-Agent: Linksys/PAP2-3.1.23(LS). Content-Length: 0. U 2011/05/21 23:40:57.187840 192.168.1.104:5060 -> 192.168.1.132:5060 SIP/2.0 200 OK - keepalive. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-56c97474;rport=5060. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: <sip:192.168.1.104>;tag=d0c53e97c845e2d86a9876bb195e3450.e05a. Call-ID: dbc8d986-9fbb95e(a)192.168.1.132. CSeq: 47 NOTIFY. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0.

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how to add RR with advertised_address in front of normal RR?
by Andrew Pogrebennyk 22 May '11

22 May '11

I've been playing with sip load-balancer in EC2 but so far haven't been able to figure out how to insert Record-Route header with advertised_address. AFAIK kamailio rr module adds 2 RR headers (one with the inbound address and one with the outbound address) when your transaction comes with one interface/port and leaves via another. In EC2 the receive socket is the same as send socket and I've had no luck trying to get kamailio insert advertised address as inbound RR and then the outbound RR, by calling record_route_preset() several times in a row too. This is a problem if lb is in front of another proxy. As I see this question has been raised on the mailing lists as far as 6 years ago - I would be surprised if no one has come up with a solution yet. So how to do record-routing in a draft-ietf-sip-record-route-fix AKA rfc5658 style on the machine with one physical interface? -- Sincerely, Andrew Pogrebennyk

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Forwarding client IP to secondary SIP server (asterisk)
by Muhammad Naseer Bhatti 22 May '11

22 May '11

Hello, I am testing Kamailio (OpenSER) as a frontend forwarding calls to asterisk. The setup is working fine and I am able to make call fine. However, Asterisk, naturally, sees calls/connections coming from Kamailio (OpenSER). Is there a way I can forward the client (calle to Kamailio (OpenSER)) IP to asterisk so asterisk can authenticate it? -B

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syntax error?
by Skyler 22 May '11

22 May '11

Hi all, I am attempting to handle NOTIFY within my script but there is a definite error in my syntax. I believe it has to do with $hdr(Event). Anyone know how I should re-write this? if (is_method("NOTIFY") && is_domain_local("$rd") && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keep-alive"); exit; } TIA

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Re: [SR-Users] syntax error?
by Skyler 22 May '11

22 May '11

Found it.will google more from now on. Sorry. if ( is_method("NOTIFY") && uri==myself && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keepalive"); xlog("L_INFO", "$hdr(Event) detected - ok sent\n"); exit; } From: Skyler [mailto:skchopperguy@gmail.com] Sent: Saturday, May 21, 2011 7:40 PM To: 'sr-users(a)lists.sip-router.org' Subject: syntax error? Hi all, I am attempting to handle NOTIFY within my script but there is a definite error in my syntax. I believe it has to do with $hdr(Event). Anyone know how I should re-write this? if (is_method("NOTIFY") && is_domain_local("$rd") && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keep-alive"); exit; } TIA

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Re: [SR-Users] Modules directory missing?
by Skyler 22 May '11

22 May '11

Found it. /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ From: Skyler [mailto:skchopperguy@gmail.com] Sent: Saturday, May 21, 2011 3:29 PM To: 'sr-users(a)lists.sip-router.org' Subject: Modules directory missing? Hello, I am new to Kamailio and have installed via rpm repository on CentOS 5.5 Where is the modules directory located? I cannot load any modules and Kamailio will not start. I've checked these locations but they don't exist. Is there another package I need to install or where did I go wrong? /usr/local/lib/kamailio/modules_k/ /usr/local/lib/kamailio/modules/

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