- sr-users - kamailio.org

[Kamailio-Users] [OT] about scalability, experiences, demands
by Henning Westerholt 14 Nov '08

14 Nov '08

Hello all, recently some statement came to my attention that "there is a common consent that the current design/architecture of [..] OpenSER (inherited from SER) is no longer able to deliver and to meet the present requirements and demands". I don't want to argue that much about this opinion, in fact the demands to a Voice over IP solution depends very much on the certain setup. But i want to share some details from my experiences in developing and operating a big VoIP infrastructure here at 1&1. We've about 2 million customers on our platform, that uses over 5 million individual numbers and terminate about 1 billion minutes per month. We're able to provide a good service with the actual architecture of OpenSER without any real problems. Of course there is always some room for improvements, but so far the main challenges we faced were not in the scalability or performance areas. More important issues are e.g. the inherent complexity of the SIP protocol and the maintainance of a good quality assurance and integration process. We started some years ago with OpenSER 0.9.5, which we then extended a lot in house. For example we implemented more than 25 own modules, a own path implementation, did a lot of bug fixing and workarounds for certain problems we've found. We're able to reduce this amount of proprietary code a lot in the past, because of progress in the OpenSER development, intregration of our "key" modules and a lot of other contributions. We're using now something between OpenSER 1.3 and Kamailio 1.4 with only a few private extensions. So in my opinion the actual design of our server is not "[..] an inevitable dead-end that needs to be avoided.". I rather think that we'll be able with continuing improvements to tackle the upcoming challenges well, especially as we will work together in the future with the SER developers in improving important areas of this software. But this is just my personal opinion, everybody is of course free to have their own position. With best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Matthias Ehrlich, Thomas Gottschlich, Robert Hoffmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen - Aufsichtsratsvorsitzender: Michael Scheeren Amtsgericht Montabaur / HRB 6484

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Re: [Kamailio-Users] Problems with counter $var(i)
by ingdavidcespedes＠cable.net.co 14 Nov '08

14 Nov '08

Ok, now I understand. So the answer is, that you can't do that with script variables (http://www.kamailio.net/dokuwiki/doku.php/pseudovariables:1.4.x#script_vari…) there it says: Note: A script variable persists over the OpenSER process in which it was initialized, so be sure of giving it a new value before reading it or you'll get the value asigned in any other previous message processed by the same OpenSER process (pid). ----- Mensaje original ----- De: Ricardo Martinez <rmartinez(a)redvoiss.net> Fecha: Jueves, Noviembre 13, 2008 2:07 pm Asunto: Re: [Kamailio-Users] Problems with counter $var(i) > Hi David. > Yes, that exactly what i want, set the variable to 0 in the main > route,so every call that arrive it start the process again... > The problem is that in the failure_route the variable keeps the > value..and is not resetting to 0 > > Ricardo.- > > -----Mensaje original----- > De: ingdavidcespedes(a)cable.net.co [ingdavidcespedes(a)cable.net.co] > > Enviado el: jueves, 13 de noviembre de 2008 16:15 > Para: Ricardo Martinez > CC: users(a)lists.kamailio.org > Asunto: Re: [Kamailio-Users] Problems with counter $var(i) > > Hi > > In which place of your main route are you writing var(i)=0. Maybe > that'sthe problem, each time you enter in the main route you are > setting the i > variable to 0. > > David Cespedes > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] Problems with counter $var(i)
by ingdavidcespedes＠cable.net.co 13 Nov '08

13 Nov '08

Hi In which place of your main route are you writing var(i)=0. Maybe that's the problem, each time you enter in the main route you are setting the i variable to 0. David Cespedes Hello I realized that the counter $var(i) in the failure_route[1] seems not to be resetting to 0 when the 2nd , 3th ..etc call arrive. So for the second call I have the counter 0 in the main route, the route(3), route(1) but in the failure_route is set to 3 (the last values from the last call) I keep trying to figure it out why this is happening.. Anyone? Thanks, Ricardo.- De: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] En nombre de Ricardo Martinez Enviado el: jueves, 13 de noviembre de 2008 11:46 Para: users(a)lists.kamailio.org Asunto: [Kamailio-Users] Problems with counter $var(i) Sorry ¡ i forgot to put the subject. Hello list. I'm having some problems trying to build a counter for my kamailio.cfg file. This is what I have.. main() ... load avp from radius} var(i)=0; route(3); ... route[1] { t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; xlog("L_INFO","[$ci] $rm t_relay a $ru"); exit; } route(3) ... $var(nr)=$avp(s:caller_nr); If ( var(i) < $var(nr) ); $var(i) = $var(i) + 1; route(1); else exit; failure_route[1] { xlog("L_INFO", "[$ci] $rm:$ru failure_route\n"); if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled en failure_route\n"); exit; } if (!next_gw()) { xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); route(3); } else { xlog("L_INFO", "XXXXX"); } t_on_failure("1"); t_relay(); } After I restart the service the first time a call arrives the counter seems to go OK, the counter goes from 0 to 2 (if the caller_nr=3). But the second time the $var(i) at the main and route(3) it's = 0, but in the failure_route it keeps the value = 3. What I'm doing wrong? Can someone help me here? Thanks Ricardo Martinez.- _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Using other media gateways
by Ruchir 13 Nov '08

13 Nov '08

I have a question. We can use media proxy to handle media through media proxies. I was wondering if it is possible to use any media servers for rtp by just specifying ip address of the media gateway.

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[Kamailio-Users] help understanding kamailio.cfg and processing call flow
by Fabian Borot 13 Nov '08

13 Nov '08

hello I have been struggling to get the grasp on kamailio for weeks now. My prob is not understanding the call flow but how to translate it using the functions provided by the modules and the core. I do not have clear the scope or reach of several functions,provided by either the core or the modules. Example: Register with Auth: 1- user sends REGISTER request w/o the encripted credentials 2- kamailio answers with 401 and sends the challenge 3- the uac sends the REGISTER again, this time with the right credentials and after verifying kamailio sends 200 OK 4- if the user is behind a NAT we can fix the contact info before saving it to the location db. With the help of the forum and the tutorials I have found in the internet I have been able to accomplish it with these commands: route{ if (uri==myself) { if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info.\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("1")){ xlog("L_INFO","mylog: REGISTER-Device behind NAT.\n"); fix_nated_register(); } force_rport(); save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } route(1); } route[1] { if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } but here come the first questions: 1- when the uac sends the first REGISTER, the function www_challenge() is the one that actually sends the "401" message? or the "exit" just exit out the "if" block and the script will continue with route(1) which in turns executes "t_relay()? ++from the doc, www_challenge() function++++ The function challenges a user agent. It will generate a WWW-Authorize header field containing a digest challenge, it will put the header field into a response generated from the request the server is processing and send the reply. +++++++++++++++++++++++++++++++++ 2- the "exit" after the save("location") function I assume would stop the execution of the script and at that point kamailio responds with "200 ok"? or just exit out the "if" block, the script then reaches the "route(1)" section and the "t_relay()" sends the "200 ok"? ++ from the doc, core-cookbook, exit() function+++++++ Stop the execution of the configuration script – it has the same behaviour as return(0). It does not affect the implicit action to be taken after script execution. ++++++++++++++++++++++++++++ 3- the "t_relay" function: does that function creates and sends the approriate response for each request? ex1: an INVITE is received, does t_relay() "know" that it can send "100 trying" automatically? ex2: a "200 OK" is received as a response to an INVITE, does t_relay knows that it has to send an "ACK"? ++++++from the doc, t_relay() funcion ++++++++++ >From user's perspective, the major function is t_relay(). It setup transaction state, absorb retransmissions from upstream, generate downstream retransmissions and correlate replies to requests. +++++++++++++++++++++++++++++ 4- why do I see ";" even after a "}", is it needed sometimes? 5- is "exit" the same as "exit()"? These may sound like dumb questions for some of the experienced users, but for a newbie they are not. Also, if this sounds like " hey, code my config file for me" I assure you that that is not the intended purpose. I would appreciate any help and thank everybody in advance. Fabian _________________________________________________________________ Color coding for safety: Windows Live Hotmail alerts you to suspicious email. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…

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[Serusers] Fw: SER COLP & COLR implementation
by Erman ÇAKMAK 13 Nov '08

13 Nov '08

Hello all, I could not find any information about SER whether it supports the COLP & COLR call features or not. Does anyone have an idea about it? Regards, Erman Cakmak

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[Kamailio-Users] 488 Not acceptable here
by Ricky Gutierrez 13 Nov '08

13 Nov '08

Hi list, I am giving him returned script from openser trying to find the problem to this message " 488 acceptable Not here" , I have my openser integrated recently with asterisk and the voicemail recently adds rtpproxy to solve problems of nat and was almost a success, some details to improve, but after adding the rtpproxy when I call to UAC and it does not answer the call on the telephone shows a 488 message to me No Aceptable, it does not jump to the voicemai.. I have 2 networks cards in my server, and looking at the sip log, I see twice in the sdp the interfaz it public any idea as solving this problem? regards rickygm # U +1.675624 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:u120@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1> Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes> Supported: replaces, timer, path Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Grandstream GXV3000 1.1.3.14 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 604 P-hint: inbound->inbound P-hint: Route[20]: Rtpproxy P-hint: Route[20]: Rtpproxy v=0 o=119 8000 8001 IN IP4 192.168.10.28 s=SIP Call c=IN IP4 192.168.1.64192.168.1.64 t=0 0 m=audio 3504835048 RTP/AVP 18 4 3 2 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 3505035050 RTP/AVP 99 34 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4CXID= a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2 MaxBR=1280 a=framerate:20 a=nortpproxy:yes a=nortpproxy:yes # U +0.000052 192.168.10.1:5060 -> 192.168.10.27:5060 CANCEL sip:120@192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1> CSeq: 12216 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000411 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000120 192.168.10.1:5060 -> 192.168.10.1:5070 ACK sip:u120@192.168.10.1:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1>;tag=as4f1165d5 CSeq: 12216 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000144 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000787 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 CANCEL User-Agent: Grandstream GXV3000 1.1.3.14 Supported: replaces, timer, 100rel, path Content-Length: 0 # U +0.000544 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Grandstream GXV3000 1.1.3.14 Content-Length: 0 # U +0.000067 192.168.10.1:5060 -> 192.168.10.27:5060 ACK sip:120@192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 CSeq: 12216 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.001390 192.168.10.28:5060 -> 192.168.10.1:5060 ACK sip:120@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Contact: <sip:119@192.168.10.28:5060> Proxy-Authorization: Digest username="119", realm="192.168.10.1", algorithm=MD5, uri="sip:120@192.168.10.1", nonce="4919edaf0a873106bfd0b18f347709360f00f7eb", response="a5a26c257b491718def2b43488a39853" Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 ACK User-Agent: Grandstream GXV3000 1.1.3.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.497023 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.999813 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact exit 67 received, 0 dropped

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[Kamailio-Users] issues with rtpproxy and kamailio
by Fabian Borot 13 Nov '08

13 Nov '08

I am trying to set up rtpproxy with kamailio and are having these 2 issues: 1- the BYEs are not relayed to the other party, I have to hang up the call at both ends 2- kamailio receives the INVITE from the calling party [behind a nat] and relays it to the called party but the connection information in the SDP has to IPs [the source IP and the RTPproxy IP together c=IN IP4 204.9.239.6765.111.185.187 user 100 is behind a NAT and user 101 is not but I want to force the RTPs to go thru the rtpproxy server [ in the same IP as kamailio] the good and weird thing is that there is audio both ways. this is my config file: if (loose_route()) { # mark routing logic in request xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("1")){ xlog("L_INFO","mylog: REGISTER-Device behind NAT.\n"); fix_nated_register(); } force_rport(); save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } else if (method=="INVITE") { xlog("L_INFO","mylog: Processing INVITE.\n"); if (!proxy_authorize("","subscriber")) { xlog("L_INFO","mylog: INVITE came without auth, sending challenge.\n"); proxy_challenge("","0"); exit; }; consume_credentials(); if (nat_uac_test("1")){ xlog("L_INFO","mylog: INVITE-Device behind NAT.\n"); #fix_nated_contact(); fix_nated_sdp("10"); } force_rtp_proxy(); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } onreply_route[1] { force_rtp_proxy("","65.111.185.187"); } thank you in advance Fabian _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Kamailio-Users] problem with fix_nated_register() function
by Fabian Borot 13 Nov '08

13 Nov '08

Hello, I am having probs with the fix_nated_register() function, I still see the private IP, even thought the nat_uac_test(31) shows the user behind a NAT This is the REGISTER section in kamailio.cfg: if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("31")){ xlog("L_INFO","mylog: Device behind NAT.\n"); fix_nated_register(); force_rport(); } save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; this is the debug log [level 6], it shows the via with a public IP Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: mylog: starting to process REGISTER.Info: [100]. Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_nonce: comparing [491a03a9000000041ba5231fd93870c44415f01d1a0d1963] and [491a03a9000000041ba5231fd93870c44415f01d1a0d1963] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_new_result: allocate 28 bytes for result set at 0x81ab8a0 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_allocate_columns: allocate 4 bytes for result names at 0x81a7288 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x81a7258 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x81a7268 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x81a7268)[0]=[password] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_convert_rows: allocate 8 bytes for rows at 0x81a7278 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_convert_row: allocate 20 bytes for row values at 0x81ab880 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_str2val: converting STRING [100] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth_db:get_ha1: HA1 string calculated: e8f9a8a478e49a182e9cceeb3e43277f Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_response: our result = '84a320db7e04e3ddd3ef84a5173187cd' Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_response: authorization is OK Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:post_auth: nonce index= 4 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing 1 columns Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x81a7268 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing result names at 0x81a7288 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing result types at 0x81a7258 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_rows: freeing 1 rows Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_row: freeing row values at 0x81ab880 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_rows: freeing rows at 0x81a7278 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_result: freeing result set at 0x81ab8a0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:check_via_address: params 204.9.239.67, 192.168.1.108, 0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: mylog: Device behind NAT. Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=8000000 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:registrar:build_contact: created Contact HF: Contact: <sip:100@192.168.1.108 7566;rinstance=89990b40ab560825>;expires=3341, <sip:100@192.168.1.108:44568;rinstance=f0942ae1a596ea83>;expires=3597 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:check_via_address: params 204.9.239.67, 192.168.1.108, 0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: mylog: save-location successful. Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:destroy_avp_list: destroying list 0xb60e11c0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:receive_msg: cleaning up and this is how the AOR is stored in the location db [ private IP] [root@sp1094a kamailio]# ../../sbin/kamctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: location table=512 records=1 max_slot=1 AOR:: 100 Contact:: sip:100@192.168.1.108:44568;rinstance=f0942ae1a596ea83 Q= Expires:: 3365 Callid:: 7a32d234b04d4b69M2IzODRhODkwM2M5NTM0MzUwMGM4ODY3NTgzYzFjMmE. Cseq:: 2 User-agent:: eyeBeam release 1003l stamp 30937 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:65.111.185.187:5060 Methods:: 5951 FIFO command was: :ul_dump:openser_receiver_14270 [root@sp1094a kamailio]# Am I missing something? Thanks in advance Fabian _________________________________________________________________ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_11…

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[Kamailio-Users] Tomcat won't start serMyAdmin - SEVERE: Error listenerStart
by rjreb 13 Nov '08

13 Nov '08

I'm trying to install serMyAdmin but it refuses to start. In the logs I see the following: Nov 12, 2008 2:25:31 AM org.apache.catalina.startup.HostConfig deployWAR INFO: Deploying web application archive serMyAdmin.war Nov 12, 2008 2:25:34 AM org.apache.catalina.core.StandardContext start SEVERE: Error listenerStart Nov 12, 2008 2:25:34 AM org.apache.catalina.core.StandardContext start SEVERE: Context [/serMyAdmin] startup failed due to previous errors I've been searching for this for a day now and I can't find where anyone else is having this problem. Any suggestions? Debian Etch Sun Java 1.5.0-16-3 Tomcat 6.0.16 -- View this message in context: http://www.nabble.com/Tomcat-won%27t-start-serMyAdmin---SEVERE%3A-Error-lis… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Serusers] Is msilo module necessary to make IM work in SER? SER doesn't forward MESSAGE
by lin9k＠yahoo.com 12 Nov '08

12 Nov '08

Hi, I used two IM (SIMPLE based) clients to communicate through SER. However, I could only see clients kept sending MESSAGE to SER but SER never forwarded them to the peer. I was not using msilo. I am not sure if that's the cause that SER didn't forward to the other end. Or, do I need any special configuration in ser.cfg, or any different version of SER? I was using ser-0.9.6. Thanks in advance! Ling

2 2

[Serusers] compiling ser from source in ubuntu
by amar mahmoud 12 Nov '08

12 Nov '08

Hi, when i compile SER from the debian packages embedded with ubuntu it works fine, but i want to compile from the source because i want to add one module i have invented. my problem is : when i compile from source mysql module does not get compiled, i have tried to change the Makefile and delete mysql module from those excluded modules, also i have tried to compile it alone with the command: make modules=modules/mysql modules but still there is problem. does any one met that before. thanks Amar _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

2 1

[Kamailio-Users] Dialog profile persistence - DB vs. memory.
by Alex Balashov 12 Nov '08

12 Nov '08

Greetings, Is some of the dialog profile information kept in in-memory structures inside the Kamailio instance even when one is using a database? I am using the dialog module in an HA configuration and fail over the database and proxy between two machines. The database and the proxy are both started on the secondary node during failover, and the database uses on-disk data and indices on a shared filesystem. What I am finding is that if I initiate some dialogs on the primary and then trigger the failover, the proxy on the secondary node is not aware of them. I see the entries for them in the database table from the secondary node - they're definitely there, and I am using db_mode 1 (real-time). However, the profile sizes are all showing as 0. I have also tried it with different match_modes with the same results whether I am using the RR cookie or SIP attributes purely. It's rather important that I get this working because I require dialog tracking for secure loose-routing of subsequent (non-initial) in-dialog requests and for concurrent call limit control. I was imagining that storing all the dialog info in the database would create the necessary persistence layer for another proxy instance to have the exact same profile information as the first. Is this a bug, or is it supposed to work this way? Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] question about register
by Fabian Borot 12 Nov '08

12 Nov '08

Hello I am having an issue handling UN-REGISTERs. This is my config file: route{ xlog("L_INFO", "mylog: starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","mylog: Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_INFO","mylog: Message too big.\n"); exit; }; if (!method=="REGISTER") { xlog("L_INFO","mylog: Recording Route. Method: [$rm].\n"); record_route(); } if (loose_route()) { # mark routing logic in request xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER" && (!search("expires=0"))) { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } else if (method=="INVITE") { xlog("L_INFO","mylog: Processing INVITE.\n"); if (!proxy_authorize("","subscriber")) { xlog("L_INFO","mylog: INVITE came without auth, sending challenge.\n"); proxy_challenge("","0"); exit; }; consume_credentials(); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } I added the "&& (!search("expires=0"))" to make sure that only the REGISTER requests will be processed by the next lines and the AOR will be saved in those cases. The X-lite sends "expires=0" in the Contact HF when it is unregistering. This is how the processing of a REGISTER looks like as per my code: Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting to process REGISTER.Info: [<null>]. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: REGISTER came without auth, sending challenge. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting to process REGISTER.Info: [101]. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: save-location successful. and when the phone UN-REGISTERs this is what I see: Nov 10 12:33:17 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:17 sp1094a ../../sbin/kamailio[26069]: mylog: lookup-location failed, sending 404 Not Found. Method: [REGISTER]. My question is, do I need to use the lookup section for REGISTERs and UN-REGISTERs? If I comment out that part this is how the log looks like: #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); # exit; #}; log: the 1st part [REGISTER] looks the same [root@sp1094a kamailio]# tail -f /var/log/kamailio_logs | grep mylog Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting to process REGISTER.Info: [<null>]. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: REGISTER came without auth, sending challenge. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting to process REGISTER.Info: [101]. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: save-location successful. and this is what the log shows when the phone is UN-REGISTERING Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Message too big. [root@sp1094a kamailio]# Regards Fabian _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

2 4

[Kamailio-Users] sip router meeting in Karlsruhe
by Daniel-Constantin Mierla 12 Nov '08

12 Nov '08

Hello, because of taking several days off, a complete report might show up a bit later, apologizes for that. Now a (short) summary... First, many thanks thanks to 1&1 for hosting the event and everyone that participated in such short notice. There were representatives from 10 companies: - 1&1 - FhG Fokus - Telio - Asipto - iptelorg/Tekelec - Voztelecom - Iptego - Itsyscom - Longphone - Basis AudioNet Some pictures should be published in the near future. After going in short introduction and presentation of the goals from the point of view of each project, we focused on: - identification of potential points of conflicts and how to get to a resolution in such case - code integration for common layer of the first phase - future development and proposals of new features - management of the larger eco-system that includes related projects and business entities I will send several emails detailing what was discussed and proposed there in few days. meanwhile, the outlines: - it is hard to avoid conflicts just by some clear and strict rules, so the common sense should lead the collaboration and discussions - GIT repository should be up in several days so the work can start, with a time line of 2-3 months from now to get core and tm in a very good shape of integration - another meeting shall be set in about 3 months time, to allow enough time for people to be able to attend, adjust the development and look more deep at the future. While the a lot of focus in the next months will be on integration, development of new features won't stop -- for examples, steps to a partial asynchronous processing are undertaking, couple of new modules are planned for release, several other modules to bring new functionalities - we should encourage and promote the development of related applications, like web interfaces, management tools, applications servers -- the add value for community and business Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] irc devel meeting summary
by Daniel-Constantin Mierla 11 Nov '08

11 Nov '08

Hello, sorry for delay... but better later than never, so I am trying to summarize the outcome of the IRC devel meeting from last Thursday, Nov 06, 2008. The log is available at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes-0… Regarding the 1.3.x series, we set the 2oth of November as the date for 1.3.4 release. There are several fixes and this branch is still used in many production environments, so it worths investing time in it. There are some updates in 1.4.x to be ported and some issues to be investigated. For a new 1.4.x release (the 1.4.3) we look for the beginning of December as the time frame for setting the date of the release. The 1.5.x rises in the first phase the question of a shorter time release, as we have lot of new features already implemented, see: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x Unfortunately some of the new features are not completed yet (e.g., htable, carrierroute updates, PV migration) so we will stick to 6-8 months release cycle. The next major one that will use the common layer (core and tm) from sip-router could be done in a shorter time than usual. So, beginning to mid of January should be the time to freeze the code for a release date somewhere in end of January or February. An important aspect approached during the discussions was the release management and testing management. Perhaps we need to appoint a person to manage releasing process in order to have a better synchronization of the developers. As everyone agrees, testing is very important and we try to improve it. There is a testing suite included in the source tree right now, used for static testing. The idea of a dynamic testing and testing community was introduced and hopefully we can attract people and build a group of beta testers. There is a proposal of how to implement and what to take care of, probably this one has to continue on devel and users mailing lists for a proper organizational structure. GIT became as a subject of migrating from SVN, because of its features for a distributed development environment. However, sourceforge does not have such offerings, so it is not possible now anyhow. However the sip-router.org will use it. Regarding the sip-router.org project, we focused on short term goals, acknowledging that we have several modules offering same functionality but relying on different underlaying data structures, this have to co-exists for a longer time. For the first phase, core + tm the overlapping is rather small, the two projects insisting in the past to improve different aspects of them. Unfortunately I forgot to bring up some punctual topics proposed, mainly related to tm failure route calling in case of sending error and reply handling/dropping in onreply_routes. We can continue those subjects on the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Question regarding config file operation
by Siddhardha Garige 11 Nov '08

11 Nov '08

Hello all, I have a basic question regarding config file operation. Here is my understanding. At build time bison and flex work on config file and generates cfg.tab.c. This file is equivalent to c version of config file. Make uses cfg.tab.c along with other files and loads a series of actions into action structures. These actions will be performed on messages in a predefined order. If i make a change to config file, do I have to rebuild openser? But I do see just by stopping and starting these changes get reflected in operation. How does these changes get into action structure with out bison and flex generating cfg.tab.c file and make compiling with other files? Any help in this regard is appreciated. Thanks Sid

3 2

[Serusers] Information request about IPv6
by Mary Allen (mrallen) 11 Nov '08

11 Nov '08

Hi, If somebody is using SERi as a Proxy with IPv6, would you please let me know the modifications/additions needed to have it working with the IPv6 and IPv4. We have three type of trunks to the Proxy (IPv4, IPv6 and Dual). IPv4 and IPv6 are configured for the machine. For the Config file do we just have to add another line for listening on one for the IPv4 and another for the IPv6? and for the routes based on the phone number I will have to entered one for an IPv6. Will that be all? Thanks, Mary

3 2

[Kamailio-Users] OpenSER HA example with vrrp/ucarp?
by sergejf 11 Nov '08

11 Nov '08

We would like to set up two OpenSER 1.4.x servers in a failover configuration. The active (primary) server has two main roles: 1) load balancing SIP INVITEs between SBCs from various carriers using the CarrierRoute module 2) collecting CDRs from all these PSTN calls using the Acc module and storing them in a MySQL database for later billing reconciliation. Right now we're using a F5 Big-IP load balancer to handle the pool of two SIP proxies which receives the SIP INVITEs from our Media Servers and dispatches them to the SBCs. However, should the primary proxy fail, the secondary proxy would never receive (for example) the SIP BYE from an SBC. Solving that issue with the Big-IP seems complicated and a simpler vrrp/ucarp solution looks attractive. Any (hands-on) advice on how to set up vrrp/ucarp to advertise a single virtual IP address, then detect that the active SIP proxy has gone down and switch to the secondary IP, would be most welcome. We're using Centos 5.x (64 bit) on our servers. Thanks a lot. Serge -- View this message in context: http://www.nabble.com/OpenSER-HA-example-with-vrrp-ucarp--tp20443679p204436… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si p EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 11 6.68.141.251:5060 "Noisy feedback tells: pid=16275 req_src_ip= 203.189.8.14 9 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:SERVER-IP vi a_cnt==1"....

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Re: [Kamailio-Users] CDRTool problem
by Ricky Gutierrez 10 Nov '08

10 Nov '08

hello, I change the values inside the section db_class, but it follows me showing the error class DB_radius extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } class DB_radius2 extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } "ser_radius"=>array( "name" => "OpenSER", "class" => "CDRS_ser_radius", "table" => "radacct", # "table" => "radacct".date("Ym"), # "db_class" => array("DB_radius2","DB_radius"), "db_class" => "DB_radius", "rating" => "1", "normalizedField" => "Normalized", "UserQuotaClass" => "OpenSERQuota", "AccountsDBClass" => "DB_openser", "UserQuotaNotify" => "1", "purgeCDRsAfter" => 120, // how many days to keep the CDRs "SIPProxies" => array("10.0.0.1" => "node01", // friendly names in sip trace table "10.0.0.2" => "node02", ), "db_class_siponline" => "DB_openser", "mediaDispatcher" => "10.0.0.1" // where to get the active media sessions from MediaProxy 2.0 I am using the version of freeradius 1.1.3 and cdrtool 6.6.10 greetingss ________________________________ Sunday, November 9, 2008 4:45:43 PM Tiago Rés Wrote: Hello, I had the same error. The problem is in the section db_class in global.inc. The default config file has in that section an array with radius1 and radius2. You have to configure it properly. That problem didn't occur more, but now When I try to read the tables it always checks cdrtool.radacct instead of radius.radacct. I don't know why it happens but I think someone else had that problem. Hope I helped.

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[Serusers] Instant Messaging? Too Many Hops
by lin9k＠yahoo.com 10 Nov '08

10 Nov '08

Hi, How do I set up the instant messaging in my SER? When I use a SIP client to send a simple message the SER responds back "Too Many Hops". How do I make it work in my SER? Ling

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[Serusers] RTP Proxy used without NAT
by Marc Bechtel 10 Nov '08

10 Nov '08

Hi, we are testing the iptel.org service using the sip server provided by iptel (213.192.59.75). Sjphonev. 1.65 is used as client. We noticed a strange behavior concerning the RTP Proxy. All SIP messages pass the iptel sip proxy as record route is used. Also all RTP packets pass the sip/rtp proxy. The question bothering us is why the RTP packets are using a proxy? (sniffs show that all RTP packets are send to the proxy) To make our case clear I will explain some details: 1. We are definitely using public IP addresses. (no RFC 1918 addresses) 2. There is no NAT. 3. Sniffs show that in the first invite packet the SDP media attribute is set to nortpproxy:yes 4. In the Sip Header via section the "rport" attribute is used. (not sure why) As I understand the SER documentation RTP proxy should only be used, if the nat_uac_test function in the nathelper module returns true. This function is checking for RFC 1918 addresses and differences in IP and port values. Thanks a lot for the help. Marc

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[Kamailio-Users] branch route
by Alex R.S.M 10 Nov '08

10 Nov '08

Hi, I am trying to modify the header of both main and the branch routes, more specifically trying to modify FROM field on both routes to different values. The change on main route is fine but couldn't make the change on the branch side. if (is_method("INVITE")) { .... seturi("sip:1111111111@x.x.x.x:5060;transport=UDP"); // main route append_branch("sip:2222222222@y.y.y.y:5060"); // branch route if ( uri=~"sip:.+@x.x.x.x.*" ) { uac_replace_from("..."); // make changes on main route, works } if ( uri=~"sip:.+@y.y.y.y.*" ) { uac_replace_from("..."); // make changes on the branch, not working } .... When I move the logic for the branch into "branch_route" it would created two FROM fields. Is this a bug? Thanks, Alex _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Serusers] STUN servers?
by SIP 10 Nov '08

10 Nov '08

Until recently, we've been using the Mystun server because it was a much more stable and reliable version of a stun server than the Vovida implementation (which hasn't been developed in a LONG time). The Mystun server, however, doesn't support XOR_MAPPED_ADDRESS or some of the other new things from the draft-ietf-behave-rfc3489bis-15. I'm wondering if anyone knows of any good stun servers out there that do. N.

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[Kamailio-Users] Routing Call
by paulo leonardo 10 Nov '08

10 Nov '08

Good Morning list, i have kamailio installed in my LAN (192.168.170.101 - machine 1), and i have eyebeam (192.168.46.101 - machine 2) registred in my kamailio, when eyebeam make a call i would like to route this call to a gateway (XXX.XXX.XXX.XXX - machine 3), how can i do that? i would like to read a practical manual. thanks

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[Kamailio-Users] Core dump on writing dialog profile size into script var.
by Alex Balashov 09 Nov '08

09 Nov '08

I also have another interesting problem with the aforementioned configuration (http://pastebin.com/f28051a5). When I write the dialog profile size into an AVP, it works fine. When I write it into a script var, i.e. replace $avp(S:dlg_sz) with $var(dlg_sz), it crashes: Relaying INVITE from 210.23.22.23 to sip:2122222322@210.23.22.23:5060 [D] Added new dialog for 7709600101 [ONREPLY-ROUTE 1] Provisional reply 100 received. [ONREPLY-ROUTE 1] 200 OK received for 7709600101 [ONREPLY-ROUTE 1] INVITE/200 is part of dialog for 7709600101 Segmentation fault (core dumped) GDB reveals: Program received signal SIGSEGV, Segmentation fault. 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6 (gdb) where #0 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6 #1 0x080a4a07 in set_var_value (var=0x8188d98, value=0xbfae5f28, flags=<value o at script_var.c:122 #2 0xb7d6c4fe in w_get_profile_size (msg=0x81902b8, profile=0xb5ede180 "\034??\ value=0x818b6b8 "@?\030\b", result=0x818b6f8 "N") at dialog.c:668 #3 0x08054f15 in do_action (a=0x818b850, msg=0x81902b8) at action.c:850 #4 0x08053ed2 in run_action_list (a=0x818b5c8, msg=0x81902b8) at action.c:138 #5 0x08056365 in do_action (a=0x818bab8, msg=0x81902b8) at action.c:722 #6 0x08053ed2 in run_action_list (a=0x818b178, msg=0x81902b8) at action.c:138 #7 0x080572c2 in run_top_route (a=0x818b178, msg=0x81902b8) at action.c:118 #8 0xb7de3a64 in reply_received (p_msg=0x81902b8) at t_reply.c:1361 #9 0x08064793 in forward_reply (msg=0x81902b8) at forward.c:507 #10 0x08090d5b in receive_msg ( buf=0x81600e0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 215.22.22.22;branch=z9hG4 08.52.173.18\r\nVia: SIP/2.0/UDP 198.225.86.10:5060;branch=z9hG4bKragjo0207gm0dc p:208.52."..., len=814, rcv_info=0xbfae65d4) at receive.c:203 #11 0x080cccfb in udp_rcv_loop () at udp_server.c:449 #12 0x0806b78d in main (argc=1, argv=0xbfae6764) at main.c:693 It would seem to me that there is some sort of buffer overflow issue that results in the garbage seen above. Not sure that it makes a difference, but the glibc being linked against is a Xen-safe one that disables TLS functionality. This is all running inside a Xen DomU. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Serusers] Assistance with SER setup
by Scott Seltzer 09 Nov '08

09 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com/> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

2 1

[Kamailio-Users] CDRTool problem
by Ricky Gutierrez 09 Nov '08

09 Nov '08

Hi forum, I am trying to configure cdrtool with openser and radius, but in the administration console it shows me this message: "Error instantiating db class: ". I have created the radius database and I see that the openser with the I modulate ACC, he writes data in the chart radacct any suggestion?, something that me this forgetting the file global.inc? best regards rickygm

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[Kamailio-Users] About 'avp_check' and strings formed with pseudo-variables
by Iñaki Baz Castillo 08 Nov '08

08 Nov '08

Hi, I've stored in "usr_preferences" some entries with: - attribute: 'bla' - value: 'alice(a)domain.org' <--- With no "sip:" in front. I use 'avp_db_load' to get that value and compare using 'avp_check' with the To uri without protocol (without "sip:"). I get this behaviour by doing: -------------------------- if avp_db_load("$fu/uri", "$avp(s:bla)") { $var(To) = $tU + '@' + $td; if avp_check("$avp(s:bla)","eq/$var(To)/gi") { [...] -------------------------- But I would like to avoid using a variable $var(To), something as: -------------------------- if avp_db_load("$fu/uri", "$avp(s:bla)") { if avp_check("$avp(s:bla)","eq/$tU + '@' + $td/gi") { # or: if avp_check("$avp(s:bla)","eq/$tU@$td/gi") { [...] -------------------------- Unfortunatelly it doesn't work, it's not valid. Do I really need to use a variable (or AVP) for "avp_check"? Thanks. -- Iñaki Baz Castillo

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[Kamailio-Users] Openser configured as outbound proxy - retransmissions issue.
by Siddhardha Garige 08 Nov '08

08 Nov '08

Hello all, I am working on configuring openser as a outbound proxy. User agents register to service provider through openser outbound proxy. I am experiencing issues with retransmissions. Case 1. UA sends an INVITE to Proxy and proxy is down. Openser sends 100 Trying to UA and retransmits INVITE messages to proxy. After 8 retry attempts sends 408 Request time out to handset. All as expected. Case 2: Openser received INVITE for UA from proxy. UA is down. Openser sent 100 Trying to proxy and routes reqest to UA. No response from UA and ** Openser did not send any retransmissions of INVITE to UA**. After a while CANCLE arrived at openser and cleared transaction. why openser does retransmissions in Case 1 and why not in case 2? Can openser proxy be operated in stateful outbound proxy to take care of retransmissions? I am using default openser config file. Is there any thing I am doing wrong? Any help is greatly appreciated. Thanks Sid

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[Kamailio-Users] [Fwd: Re: [Sip-implementors] REGISTER R-URI with port parameter]
by Alex Balashov 07 Nov '08

07 Nov '08

Does Kamailio match 'uri == myself' if the domain is the same as the canonical hostnames/aliases but the port is different than the one Kamailio knows itself to be listening on? -------- Original Message -------- Subject: Re: [Sip-implementors] REGISTER R-URI with port parameter Date: Tue, 28 Oct 2008 16:46:04 -0500 From: Dale.Worley(a)comcast.net To: sip-implementors(a)lists.cs.columbia.edu References: <00D42150952F70458C66072322F7FE2502EDA940(a)saturn2.aculab.com> <200810241652.m9OGqx6s027493(a)dragon.ariadne.com> <200810241902.43581.ibc(a)aliax.net> From: =?iso-8859-1?q?I=F1aki_Baz_Castillo?= <ibc(a)aliax.net> El Viernes, 24 de Octubre de 2008, Dale.Worley(a)comcast.net escribió: > sip:xxx@host:5060 > sip:xxx@host:15060 > sip:xxx@host:25060 > sip:xxx@host:35060 > > In that case, the request-URI of the REGISTER needs to contain the > proper port number. Well, not totally needed. A UA could construct a REGISTER like this: REGISTER sip:xxx@host SIP/2.0 and sent it to host:15060. You can't depend on that working -- It's never been settled what must happen if a SIP message arrives at a UAS which isn't the UAS that the RFC 3263 rules would send it to. It's possible that the UAS will forward it based on the request-URI. It's safer to include the port, if the destination port is not 5060. For example Twinkle does it if you set the registrar in a port different than 5060. I wouldn't depend on that working. Dale _______________________________________________ Sip-implementors mailing list Sip-implementors(a)lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Passing REGISTERs up stream
by Daniel Corbe 07 Nov '08

07 Nov '08

Guys, When attempting to pass REGISTER requests up stream I get the following error messages on the console (below). This also brings up the question of "what is the best way to simply pass REGISTER requests on to an upstream registrar?" Thanks for the help. -Daniel REGISTER: NATed client, enabling NAT Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed Oct 29 18:03:59 [48175] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 401 Here's the relevant parts of my ser config; in route[1] route[1] { ... if (method == "REGISTER") { route(2); } route(1); } route[2] { # Check to see if the UAC is trying to UNREGISTER. If not test # for NAT. If NAT is present, we mark it as such before we save() # the location, that way flag 6 is always set for NATed UAs. if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { xlog("L_NOTICE", "REGISTER: NATed client, enabling NAT\n"); setflag(6); fix_nated_register(); force_rport(); }; # We must handle replies for registrations, for caching and location # tracking purposes. t_on_reply("1"); t_on_failure("1"); # Check for digest #if (radius_www_authorize("")) #{ # xlog("L_NOTICE", "REGISTER: No Digest, sending challenge\n"); # www_challenge("", "0"); # exit; #} # Digest was good if we get here #save("location"); # Relay register to porta t_relay("216.151.143.69"); } onreply_route[2] { if (status =~ "2[0-9][0-9]") { # save("location"); }; }

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[Serusers] Problem with SIP Identity
by jerome.herve＠orange-ftgroup.com 07 Nov '08

07 Nov '08

> Hello everyone, > I'm currently working on the implementation of RFC4474 in SER. > I didn't manage to set up the auth_identity module in SER. > Is there someone able to tell me what do I have to write in ser.cfg to make this possible? > An example of an entire ser.cfg will be great. > I hope you will help me. > Thanks & regards, > > Jérôme HERVE > tél. 02 96 05 27 41 > mob. 06 76 15 18 49 > >

3 2

[Serusers] Problem with SIP Identity
by jerome.herve＠orange-ftgroup.com 07 Nov '08

07 Nov '08

Hello everyone, I'm currently working on the implementation of RFC4474 in SER. I didn't manage to set up the auth_identity module in SER. Is there someone able to tell me what do I have to write in ser.cfg to make this possible? An example of an entire ser.cfg will be great. I hope you will help me. Thanks & regards, Jérôme HERVE tél. 02 96 05 27 41 mob. 06 76 15 18 49

2 1

[Kamailio-Users] $du has no effect in failure route
by jh＠tutpro.com 07 Nov '08

07 Nov '08

i have in failure route: t_on_branch("0"); $du = "sip:127.0.0.1:5070"; xlog("L_INFO", "Pseudo variable du has value <$du>\n"); t_relay("0x01"); and i get to syslog: Nov 6 16:33:15 test /usr/sbin/kamailio[22539]: Pseudo variable du has value <sip:127.0.0.1:5070> but wireshark shows that the invite is sent according to uri host of the branch added in failure route and not to 127.0.0.1:5070. is this a bug or is it so that $du is not supposed to have any effect in failure route? -- juha

3 3

[Kamailio-Users] Kamailio + eyebeam + TLS
by paulo leonardo 07 Nov '08

07 Nov '08

Hi list, I have a litle problem, i would like to setup eyebeam with kamailio using TLS, but i always get *Registration error: 503 - certificate validation failure* =( !! I read the README of TLS. How certificate (.crt) i need to install in eyebeam? # kamailio -V ------------------------------------------------------------------------------------------- version: kamailio 1.4.1-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP,* USE_TLS*, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 4623 2008-08-07 07:31:30Z miconda $ main.c compiled on 16:31:23 Oct 24 2008 with gcc 4.2.3 ------------------------------------------------------------------------------------------ disable_tls = 0 listen = tls:192.168.170.101:5061 tls_port_no = 5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" ------------------------------------------------------------------------------------------------------------ thanks list!

2 1

Re: [Kamailio-Users] Remote-Party-ID header
by ingdavidcespedes＠cable.net.co 07 Nov '08

07 Nov '08

2 1

Re: [Kamailio-Users] CANCEL and 487 Request Terminated
by ingdavidcespedes＠cable.net.co 06 Nov '08

06 Nov '08

I think that's the way it should be. First it receive 487 from SBC and immediately it responds it to SBC with the ACK. Then it passes the 487 to the UAC and receive the ACK, that's all. It doesn't have to forward that last ACK to the SBC. ----- Mensaje original ----- De: Krom <kyle.romulas(a)gmail.com> Fecha: Jueves, Noviembre 6, 2008 4:06 pm Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > My apologies for the drawing. Openser is receiving the 487 from the > SBC, ACKing back to the SBC and then passing the 487 to the UAC and > the the UAC ACK's back to Openser. At which time the error occurs > because the ACK will not be forwarded since it already ACK'd for the > 487 to the SBC. > > -Kyle > > On Thu, Nov 6, 2008 at 2:42 PM, <ingdavidcespedes(a)cable.net.co> > wrote:> The final 487 should go from openser to UAC, not from SBC > to UAC. That's why the final ACK is getting lost, because it was > allready ACK'd by openser. > > > > ----- Mensaje original ----- > > De: Krom <kyle.romulas(a)gmail.com> > > Fecha: Jueves, Noviembre 6, 2008 1:18 pm > > Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > > > >> Klaus, > >> > >> Some additional information. After viewing a trace I can see > that the > >> error is being generated from the last ACK received from the UAC. > >> OpenSER/Kamailio is ACK'ing to the 487 from the PSTN provider > and then > >> relaying the 487 to the UAC which in turn ACK's back. It is the > last>> ACK from the UAC that is getting the error on t_relay. I > can assume > >> because the ACK to the 487 was already sent by OpenSER/Kamailio > to the > >> PSTN upon receiving the initial 487. Something in the script I can > >> change to prevent this or just ignore the error and move on. :-) > >> > >> UAC OSER SBC > >> > >> CANCEL ---> | > >> > >> CANCEL ---> | > >> > >> <--- 200 canceling > >> > >> | <--- 200 OK > >> > >> | <--- 487 Request Terminated > >> > >> ACK ---> > >> > >> <--- 487 Reqest Terminated > >> > >> ACK ---> | > >> > >> > >> Regards, > >> -Kyle > >> > >> > >> On Thu, Nov 6, 2008 at 9:05 AM, Krom <kyle.romulas(a)gmail.com> > wrote:>> > Klaus, > >> > > >> > Thank you for the reply and the link. The link help in > checking my > >> > work. My script is very simple and even checking against the > >> compiled> scripted from sipwise.com the error persists. > >> > > >> > Here is my CANCEL and ACK handling. > >> > if(is_method("CANCEL") || is_method("ACK")) { > >> > if(t_check_trans()) { > >> > if(!t_relay()) sl_reply_error(); > >> > } > >> > exit; > >> > } > >> > > >> > Here is my route block for PSTN calls. > >> > route[4] > >> > { > >> > if (!load_gws()) { > >> > sl_send_reply("500", "Server Internal Error"); > >> > exit; > >> > } > >> > if (!next_gw()) { > >> > sl_send_reply("503", "Service Unavailable"); > >> > exit; > >> > } > >> > t_on_failure("2"); > >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder > >> Controller> > >> > exit; > >> > > >> > } > >> > > >> > Here is my failure route. > >> > failure_route[2] > >> > { > >> > > >> > if(t_check_status("408|500|503")) { > >> > if (!next_gw()) { > >> > t_reply("503", "Service Unavailable"); > >> > exit; > >> > } > >> > t_on_failure("2"); > >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder > >> Controller> } > >> > > >> > exit; > >> > > >> > } > >> > > >> > Any thoughts or errors you see? > >> > > >> > Regards. > >> > > >> > > >> > On Thu, Nov 6, 2008 at 3:56 AM, Klaus Darilion > >> > <klaus.mailinglists(a)pernau.at> wrote: > >> >> looks like the ACK is miss-routed. > >> >> > >> >> Usually calling t_relay() for ACK is sufficient (take a look at > >> the config > >> >> file generated with sipwise.com wizard for correct CANCEL/ACK > >> handling.>> > >> >> regards > >> >> klaus > >> >> > >> >> Krom schrieb: > >> >>> > >> >>> Et al, > >> >>> > >> >>> Version: openser-1.1.1-notls > >> >>> > >> >>> I am receiving the following after calling t_relay within the > >> default>>> route block. > >> >>> ERROR: t_newtran: transaction already in process 0xb0139638 > >> >>> WARNING:sl:sl_send_reply: I won't send a reply for ACK!! > >> >>> WARNING: script writer didn't release transaction > >> >>> > >> >>> This is happening when a downstream UAC dials outbound to the > >> PSTN and > >> >>> then terminates the call prior to the far ending answering. > >> >>> > >> >>> UAC send a CANCEL to the proxy, the proxy responds to the UAC > >> with a > >> >>> '200 canceling' and forwards the CANCEL to the PSTN provider. > >> >>> The PSTN provider sends 487 Request Terminated to the proxy and > >> the>>> proxy sends it on to the UAC. UAC responds with an ACK to > >> the proxy. > >> >>> > >> >>> Is this an issue? How do I handle this dialog withou > getting the > >> >>> t_relay error message? > >> >>> > >> >>> > >> >> > >> > > >> > > >> > > >> > -- > >> > Thanks, > >> > -- Kyle > >> > > >> > >> > >> > >> -- > >> Thanks, > >> -- Kyle > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > > > -- > Thanks, > -- Kyle >

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Re: [Kamailio-Users] CANCEL and 487 Request Terminated
by Krom 06 Nov '08

06 Nov '08

My apologies for the drawing. Openser is receiving the 487 from the SBC, ACKing back to the SBC and then passing the 487 to the UAC and the the UAC ACK's back to Openser. At which time the error occurs because the ACK will not be forwarded since it already ACK'd for the 487 to the SBC. -Kyle On Thu, Nov 6, 2008 at 2:42 PM, <ingdavidcespedes(a)cable.net.co> wrote: > The final 487 should go from openser to UAC, not from SBC to UAC. That's why the final ACK is getting lost, because it was allready ACK'd by openser. > > ----- Mensaje original ----- > De: Krom <kyle.romulas(a)gmail.com> > Fecha: Jueves, Noviembre 6, 2008 1:18 pm > Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > >> Klaus, >> >> Some additional information. After viewing a trace I can see that the >> error is being generated from the last ACK received from the UAC. >> OpenSER/Kamailio is ACK'ing to the 487 from the PSTN provider and then >> relaying the 487 to the UAC which in turn ACK's back. It is the last >> ACK from the UAC that is getting the error on t_relay. I can assume >> because the ACK to the 487 was already sent by OpenSER/Kamailio to the >> PSTN upon receiving the initial 487. Something in the script I can >> change to prevent this or just ignore the error and move on. :-) >> >> UAC OSER SBC >> >> CANCEL ---> | >> >> CANCEL ---> | >> >> <--- 200 canceling >> >> | <--- 200 OK >> >> | <--- 487 Request Terminated >> >> ACK ---> >> >> <--- 487 Reqest Terminated >> >> ACK ---> | >> >> >> Regards, >> -Kyle >> >> >> On Thu, Nov 6, 2008 at 9:05 AM, Krom <kyle.romulas(a)gmail.com> wrote: >> > Klaus, >> > >> > Thank you for the reply and the link. The link help in checking my >> > work. My script is very simple and even checking against the >> compiled> scripted from sipwise.com the error persists. >> > >> > Here is my CANCEL and ACK handling. >> > if(is_method("CANCEL") || is_method("ACK")) { >> > if(t_check_trans()) { >> > if(!t_relay()) sl_reply_error(); >> > } >> > exit; >> > } >> > >> > Here is my route block for PSTN calls. >> > route[4] >> > { >> > if (!load_gws()) { >> > sl_send_reply("500", "Server Internal Error"); >> > exit; >> > } >> > if (!next_gw()) { >> > sl_send_reply("503", "Service Unavailable"); >> > exit; >> > } >> > t_on_failure("2"); >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder >> Controller> >> > exit; >> > >> > } >> > >> > Here is my failure route. >> > failure_route[2] >> > { >> > >> > if(t_check_status("408|500|503")) { >> > if (!next_gw()) { >> > t_reply("503", "Service Unavailable"); >> > exit; >> > } >> > t_on_failure("2"); >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder >> Controller> } >> > >> > exit; >> > >> > } >> > >> > Any thoughts or errors you see? >> > >> > Regards. >> > >> > >> > On Thu, Nov 6, 2008 at 3:56 AM, Klaus Darilion >> > <klaus.mailinglists(a)pernau.at> wrote: >> >> looks like the ACK is miss-routed. >> >> >> >> Usually calling t_relay() for ACK is sufficient (take a look at >> the config >> >> file generated with sipwise.com wizard for correct CANCEL/ACK >> handling.>> >> >> regards >> >> klaus >> >> >> >> Krom schrieb: >> >>> >> >>> Et al, >> >>> >> >>> Version: openser-1.1.1-notls >> >>> >> >>> I am receiving the following after calling t_relay within the >> default>>> route block. >> >>> ERROR: t_newtran: transaction already in process 0xb0139638 >> >>> WARNING:sl:sl_send_reply: I won't send a reply for ACK!! >> >>> WARNING: script writer didn't release transaction >> >>> >> >>> This is happening when a downstream UAC dials outbound to the >> PSTN and >> >>> then terminates the call prior to the far ending answering. >> >>> >> >>> UAC send a CANCEL to the proxy, the proxy responds to the UAC >> with a >> >>> '200 canceling' and forwards the CANCEL to the PSTN provider. >> >>> The PSTN provider sends 487 Request Terminated to the proxy and >> the>>> proxy sends it on to the UAC. UAC responds with an ACK to >> the proxy. >> >>> >> >>> Is this an issue? How do I handle this dialog withou getting the >> >>> t_relay error message? >> >>> >> >>> >> >> >> > >> > >> > >> > -- >> > Thanks, >> > -- Kyle >> > >> >> >> >> -- >> Thanks, >> -- Kyle >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > > -- Thanks, -- Kyle

1 0

[Kamailio-Users] CANCEL and 487 Request Terminated
by Krom 06 Nov '08

06 Nov '08

Et al, Version: openser-1.1.1-notls I am receiving the following after calling t_relay within the default route block. ERROR: t_newtran: transaction already in process 0xb0139638 WARNING:sl:sl_send_reply: I won't send a reply for ACK!! WARNING: script writer didn't release transaction This is happening when a downstream UAC dials outbound to the PSTN and then terminates the call prior to the far ending answering. UAC send a CANCEL to the proxy, the proxy responds to the UAC with a '200 canceling' and forwards the CANCEL to the PSTN provider. The PSTN provider sends 487 Request Terminated to the proxy and the proxy sends it on to the UAC. UAC responds with an ACK to the proxy. Is this an issue? How do I handle this dialog withou getting the t_relay error message? -- Thanks, -- Kyle

2 3

[Serusers] Assistance with SER setup
by Scott Seltzer 06 Nov '08

06 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com/> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

1 0

[Serusers] FW: Support for SER setup
by Scott Seltzer 06 Nov '08

06 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

1 0

[Kamailio-Users] two endpoints
by Alex R.S.M 06 Nov '08

06 Nov '08

Hi, If ENUM lookup returns 2 destinations, does kamailio fork the message and send two INVITE? What function can be used to force kamailio to fork a message? Regards, Alex _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

2 1

[Kamailio-Users] rewriteuser with AVP
by Samuel Muller 06 Nov '08

06 Nov '08

Hello all, I've a little question : I got a username in an AVP by avp_load_radius, and I would rewrite the uri with it. The objectives is to replace the r-uri by an ipbx uri, then forward the call to it by a new branch (to not loose the phone number requested). in this case : 0123452000 is behind a spa9000. The user 0123452000 has an AVP called "ipbx", that i'm using to do some groups and permissions (plus the rewrite). actually, it does not work, i tried many ways : # load radius attributes of the callee if (!avp_load_radius("callee")) { xlog("L_INFO","-> user unknown in radius usr db : $ru"); route(15); # route PSTN } # verify the AVPs we got (caller and callee) xlog("L_INFO", " -AVP------------------------------------- "); avp_print(); xlog("L_INFO", " ----------------------------------------- "); # callee is behind an ipbx (avp ipbx) ? if (is_avp_set("$avp(s:callee_ipbx)")) { xlog("L_INFO", "-> callee is behind an ipbx : $avp(s:callee_ipbx)"); setflag(14); # flag IPBX #subst_user('/$rU$/$avp(s:callee_ipbx)/'); #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); rewriteuser($avp(s:callee_ipbx)); } # callee in usrloc ? if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } else { xlog("L_INFO","-> 480: local user not found in usrloc : $tu"); sl_send_reply("480","Temporarily Unavailable"); drop; } and in the debugs log : Nov 5 13:04:51 ser0 kamailio[2930]: -ROUTE--INBOUND-------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth Success Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_asserted_id'/0='0123452000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_ipbx'/0='spa9000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: -AVP------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cfe8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_ipbx> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<spa9000 / 7> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cf88, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123452000 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ced8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<caller_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123451011 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ce38, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: ----------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( s(a)sip.720.fr' Not found in usrloc Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params 77.246.81.162, 192.168.0.134, 0 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying list 0x7fb3d7e5d040 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up -> But, if I did it by : if (uri =~ "^sip:012345200{1}") { rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> "); } if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } it works great, and the call is ok (the ipbx forward correctly the request the phone behind him). Anyone has an idea ? thanks in advance ! -- Samuel MULLER

2 3

[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

Hi David, thank you for you answer, 2008/11/6, ingdavidcespedes(a)cable.net.co <ingdavidcespedes(a)cable.net.co>: > > Hi! > > I had the same problem (My PC was responding everything it receives through > the default gateway "eth0" ). Everybody say that it was an Asterisk bug, > but it was a problem in the routing table. Yes, I'm new to Openser but I know quite well Asterisk, and I don't think it is a bug or a misconfiguration . Type route in a terminal and check if they are the way you are thinking. If > not, add the correct route. Tell me if it solve your problem. The route table is correct, but a strange thing happens since the SIP packet is sent from the correct nic, but un the UDP header of that packet I see as source the IP of the other nic (the one that received the orinal packet) I have to investigate a little more on this aspect anyway. C. David Cespedes > > ----- Mensaje original ----- > De: Cosimo Fadda <cfadda.lists(a)gmail.com> > Fecha: Miércoles, Noviembre 5, 2008 9:36 am > Asunto: [Kamailio-Users] Openser as Proxy and Asterisk as Registrar > > > Hi everybody > > I'd like to implement this scenario: > > > > ----------- ------------------- ---------- > > ----- > > | GW/FW > > |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -- > > >PSTN----------- ------------------- | ----- > > ---------- > > |_______________________________________________| > > > > > > > > > > > > Where: > > - Openser is reachable from the outside with a public IP > > forwarded to > > eth0 private address; > > > > - Asterisk is connected to Openser using another separated LAN > > - Openser acts as proxy (for requests coming from the outside) and > > Asterisk act as registrar and gateway. > > > > > > This is the description, now the questions: > > -since opneser has two different nics, how can configure it to > > properly > > send register request to asterisk? > > This is what currently happens: > > 192.168.40.68:5060: is eth0 Openser Address > > 192.168.12.165 is eth1 Openser Address > > 192.168.12.106 is Asterisk address > > 82.187.X.X is Openser public address > > Network 192.168.40.X is not reachable from net 192.168.12.X > > > > <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 > > SIP/2.0 // 12.106 is Asterisk address > > Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: > > SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> > > Contact: "Cfadda" <sip:225@192.168.40.254:2660> > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > Expires: 1800 > > Max-Forwards: 69 > > User-Agent: X-PRO release 1105x > > Content-Length: 0 > > > > > > Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) -- > > - > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER > > request > > as basis request > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : > > 5060 > > (non-NAT) > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X > > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> > > Content-Length: 0 > > > > > > --- > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > > nonce="05f14a3e"Content-Length: 0 > > > > So asterisk sends response to the wrong interface. > > > > How can I solve this? > > > > Thanks in advance, > > > > C. > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

Hi Daniel, I'll try to use the force_send_sock function, maybe this solve the issue; btw I don't think it is a strictly Asterisk related problem, thank you for the hint, C. 2008/11/5, Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, > > On 11/05/08 16:36, Cosimo Fadda wrote: > >> Hi everybody >> I'd like to implement this scenario: >> >> ----------- ------------------- --------------- >> | GW/FW |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| >> -->PSTN >> ----------- ------------------- | --------------- >> |_______________________________________________| >> >> >> >> Where: >> - Openser is reachable from the outside with a public IP forwarded to >> eth0 private address; >> >> - Asterisk is connected to Openser using another separated LAN >> - Openser acts as proxy (for requests coming from the outside) and >> Asterisk act as registrar and gateway. >> >> >> This is the description, now the questions: >> -since opneser has two different nics, how can configure it to properly >> send register request to asterisk? >> >> > if you want to force an interface for outgoing sip requests, see > force_send_sock() function: > > http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.4.x#force_send_so… > > This is what currently happens: >> 192.168.40.68:5060: is eth0 Openser Address >> 192.168.12.165 is eth1 Openser Address >> 192.168.12.106 is Asterisk address >> 82.187.X.X is Openser public address >> Network 192.168.40.X is not reachable from net 192.168.12.X >> >> <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106SIP/2.0 // 12.106 is Asterisk address >> Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: SIP/2.0/UDP >> 192.168.40.254:2660;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> >> >;tag=3189722840 >> To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> >> Contact: "Cfadda" <sip:225@192.168.40.254:2660> >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> Expires: 1800 >> Max-Forwards: 69 >> User-Agent: X-PRO release 1105x >> Content-Length: 0 >> >> >> Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) --- >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER request as >> basis request >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : 5060 >> (non-NAT) >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to >> 192.168.40.68:5060: >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received= >> 192.168.40.68 >> Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> >> >;tag=3189722840 >> To: Cfadda <sip:225@82.187.X.X > >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> >> Content-Length: 0 >> >> >> --- >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to >> 192.168.40.68:5060: >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received= >> 192.168.40.68 >> Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 >> To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05f14a3e" >> Content-Length: 0 >> >> So asterisk sends response to the wrong interface. >> >> How can I solve this? >> >> > > Probably the right answer you will get on the asterisk mailing lists. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

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[Kamailio-Users] reminder: irc devel meeting in 1h20
by Daniel-Constantin Mierla 06 Nov '08

06 Nov '08

Hello, just a quick reminder about the irc devel meeting: - channel: #kamailio - server: irc.freenode.net - time: 15:00UTC Deatils and agenda: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] SER/Asterisk interworking mailing list
by luzango mfupe 06 Nov '08

06 Nov '08

Hi Alex,This is good news coming from you!!! Yes We Can....!!! Rgds, Lu. On Wed, Nov 5, 2008 at 7:48 PM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > > 5. SER/Asterisk interworking mailing list. (Alex Balashov) > > > Message: 5 > Date: Wed, 05 Nov 2008 12:04:32 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: [Kamailio-Users] SER/Asterisk interworking mailing list. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users(a)lists.digium.com>, Commercial and > Business-Oriented > Asterisk Discussion <asterisk-biz(a)lists.digium.com>, users > <users(a)lists.kamailio.org>, openser users < > users(a)lists.opensips.org> > Message-ID: <4911D220.1040506(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Greetings, > > As a developer and consultant who spends considerable time on projects > involving the fusion of Asterisk and products derived from the SER > ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have > found that there is a great volume of interest in this topic on the > mailing lists associated with all communities involved, but a > comparative lack of focus that results in duplicated effort and lack of > specialised response. > > This is mainly due, I think, to the fact that detailed Asterisk > experience - while common - is not a prerequisite for working with the > SER products, while for Asterisk people SER can often be a next step in > scalability and VoIP service delivery platform enhancement that they are > just getting into. And so on. There's pollution in the respective > discursive spaces; a lot of Asterisk people posting to the SER lists > ask a lot of Asterisk-specific questions in addition to any they may > have about SER which can be construed as potentially off-topic by some > members, and the opposite is true on the Asterisk lists when detailed, > involved discussion about SER occurs. > > We need to capture that discussion that exists at the overlap and is > specifically concerned with making these two systems work together, > requiring somewhat detailed and esoteric understanding of both and a > community of user support and knowledge that focuses on both of these > conceptual and product universes. > > Toward that end, I am hosting a new mailing list with this succinct > purpose, if slightly unwieldy name, and encourage all interested to > join. It is called 'SER-Asterisk-Interwork' and can be accessed for > subscription here: > > http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork > > The archives are available here: > > http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ > > You can post to the list at: > > ser-asterisk-interwork(a)lists.evaristesys.com > > It's the same GNU Mailman stuff you are already used to. > > While it could be argued that this cross-product discussion is valuable > to retain in both communities, I think there is considerable benefit to > creating a specialised mailing list that focuses specifically on this > integration path and the unique interoperation and configuration issues > it creates. I think it would be good to get some of this discussion off > of the SER and Asterisk-specific mailing lists where it has somewhat > marginal relevance at times and refocus it. If you agree and are > interested in this topic, you are invited to join the list. > > One last note: The SER/OpenSER community has been in a state of flux > recently, with OpenSER undergoing a name change to Kamailio and > subsequently seeing a fork. The incumbent Kamailio project is now > in the process of merging with the original SER project. The choice of > nomenclature for list is not meant to imply an endorsement of or > affinity for the IPTel SER project per se. It is just that right now it > serves the aim of terseness to use a common denominator, to refer to > this family of projects as the "SER ecosystem." Whether you are a SER, > OpenSER, Kamailio, or OpenSIPS user, you are part of that "SER > ecosystem." That is why the list is named what it is. > > Thank you, > > -- Alex > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 15 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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Re: [Kamailio-Users] Is it possible to break a "if" stament?
by ingdavidcespedes＠cable.net.co 06 Nov '08

06 Nov '08

----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Miércoles, Noviembre 5, 2008 6:50 pm Asunto: [Kamailio-Users] Is it possible to break a "if" stament? > Hi, is it possible to break the body of an "if"? something like: > > -------------- > if $fU == "alice" { > xlog "hello"; > break; > xlog "bye"; > } > xlog "this is after the 'if'" > -------------- > > So the result would be: > > ------- > hello > this is after the 'if' > ------- > > Is it possible? > Not, you can't do brake inside an If, only for Switch. Why don't you do if $fU == "alice" { xlog "hello"; } else{ xlog "bye"; } xlog "this is after the 'if'" > -- > Iñaki Baz Castillo > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] high processing delay
by Fabian Borot 06 Nov '08

06 Nov '08

hello All my Register request take longer than normal to process, what I mean is that an UAC sends a Register and kamailio responds almost 0.5 secs later, causing the UAC to re-transmit its request several times 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 0.469256 192.168.1.20 -> 192.168.10.10 SIP Status: 401 Unauthorized (0 bindings) 0.518263 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 0.628294 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 1.130722 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 1.269301 192.168.1.20 -> 192.168.10.10 SIP Status: 401 Unauthorized (0 bindings) 2.224942 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 2.325358 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) 3.457509 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) 4.497532 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) the cpu shows almost 100 % idle at that time and the pc were kamailio is running has nothing else running but kamailio. this is my config file: ####### Routing Logic ######## # main request routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests xlog("L_DBG", "mylog, time [$Tf] : starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_DGB","mylog, time [$Tf] : Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_DGB","mylog, time [$Tf] : Message too big.\n"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_DGB","mylog, time [$Tf] : starting to process REGISTER.Info: [$au,$ad,$ci,$ct,$cs,$rd,$si,$sp].\n"); if (!www_authorize("", "subscriber")) { xlog("L_DGB","mylog, time [$Tf] : REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; save("location"); xlog("L_DBG","mylog, time [$Tf] : save-location successful.\n"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_DGB","mylog, time [$Tf] : lookup-location failed, sending 404 Not Found.\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Kamailio-Users] Is it possible to break a "if" stament?
by Iñaki Baz Castillo 05 Nov '08

05 Nov '08

Hi, is it possible to break the body of an "if"? something like: -------------- if $fU == "alice" { xlog "hello"; break; xlog "bye"; } xlog "this is after the 'if'" -------------- So the result would be: ------- hello this is after the 'if' ------- Is it possible? -- Iñaki Baz Castillo

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Re: [Kamailio-Users] high processing delay
by Daniel-Constantin Mierla 05 Nov '08

05 Nov '08

On 11/05/08 20:15, Fabian Borot wrote: > debug was set to 9, I commented that line out and now it is fine: ok > > > 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > sip:192.168.1.20 > 0.001650 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > Unauthorized (0 bindings) > 0.191670 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > sip:192.168.1.20 > 0.192518 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > bindings) > > but how can I check about this "syslog set to asynchronous mode" ? check your syslog.conf or so, depends on you distribution. Probably the safest is to check the manual, the usual way for old versions of syslog was to put a - (dash) in front of the file for logging. Cheers, Daniel > > regards > fabian > > > > > > Date: Wed, 5 Nov 2008 19:37:35 +0200 > > From: miconda(a)gmail.com > > To: fborot(a)hotmail.com > > CC: users(a)lists.kamailio.org > > Subject: Re: [Kamailio-Users] high processing delay > > > > hello, > > > > what is the debug level? is you syslog set to asynchronous mode? > > > > Cheers, > > Daniel > > > > > > On 11/05/08 17:43, Fabian Borot wrote: > > > hello > > > > > > All my Register request take longer than normal to process, what I > > > mean is that an UAC sends a Register and kamailio responds almost 0.5 > > > secs later, causing the UAC to re-transmit its request several times > > > > > > 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 0.469256 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > > > Unauthorized (0 bindings) > > > 0.518263 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 0.628294 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 1.130722 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 1.269301 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > > > Unauthorized (0 bindings) > > > 2.224942 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 2.325358 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > 3.457509 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > 4.497532 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > > > > > > > the cpu shows almost 100 % idle at that time and the pc were kamailio > > > is running has nothing else running but kamailio. this is my > config file: > > > > > > > > > > > > ####### Routing Logic ######## > > > > > > > > > # main request routing logic > > > > > > > > > route{ > > > # initial sanity checks -- messages with > > > # max_forwards==0, or excessively long requests > > > xlog("L_DBG", "mylog, time [$Tf] : starting_main_logic.\n"); > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483","Too Many Hops"); > > > xlog("L_DGB","mylog, time [$Tf] : Too Many Hops.\n"); > > > exit; > > > }; > > > > > > if (msg:len >= 2048 ) { > > > sl_send_reply("513", "Message too big"); > > > xlog("L_DGB","mylog, time [$Tf] : Message too big.\n"); > > > exit; > > > }; > > > > > > > > > if (!method=="REGISTER") > > > record_route(); > > > > > > if (loose_route()) { > > > # mark routing logic in request > > > append_hf("P-hint: rr-enforced\r\n"); > > > route(1); > > > }; > > > > > > > > > if (uri==myself) { > > > if (method=="REGISTER") { > > > xlog("L_DGB","mylog, time [$Tf] : starting to > > > process REGISTER.Info: [$au,$ad,$ci,$ct,$cs,$rd,$si,$sp].\n"); > > > if (!www_authorize("", "subscriber")) { > > > xlog("L_DGB","mylog, time [$Tf] : REGISTER > > > came without auth, sending challenge.\n"); > > > www_challenge("", "0"); > > > exit; > > > }; > > > save("location"); > > > xlog("L_DBG","mylog, time [$Tf] : save-location successful.\n"); > > > exit; > > > }; > > > > > > > > > if (!lookup("location")) { > > > sl_send_reply("404", "Not Found"); > > > xlog("L_DGB","mylog, time [$Tf] : lookup-location > > > failed, sending 404 Not Found.\n"); > > > exit; > > > }; > > > append_hf("P-hint: usrloc applied\r\n"); > > > }; > > > route(1); > > > } > > > > > > route[1] { > > > if (!t_relay()) { > > > sl_reply_error(); > > > }; > > > exit; > > > } > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > Stay up to date on your PC, the Web, and your mobile phone with > > > Windows Live Click here > > > <http://clk.atdmt.com/MRT/go/119462413/direct/01/> > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Daniel-Constantin Mierla > > http://www.asipto.com > > > > ------------------------------------------------------------------------ > Color coding for safety: Windows Live Hotmail alerts you to suspicious > email. Sign up today. > <http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…> -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 05 Nov '08

05 Nov '08

Hi everybody I'd like to implement this scenario: ----------- ------------------- --------------- | GW/FW |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -->PSTN ----------- ------------------- | --------------- |_______________________________________________| Where: - Openser is reachable from the outside with a public IP forwarded to eth0 private address; - Asterisk is connected to Openser using another separated LAN - Openser acts as proxy (for requests coming from the outside) and Asterisk act as registrar and gateway. This is the description, now the questions: -since opneser has two different nics, how can configure it to properly send register request to asterisk? This is what currently happens: 192.168.40.68:5060: is eth0 Openser Address 192.168.12.165 is eth1 Openser Address 192.168.12.106 is Asterisk address 82.187.X.X is Openser public address Network 192.168.40.X is not reachable from net 192.168.12.X <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 SIP/2.0 // 12.106 is Asterisk address Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.90.68>;tag=3189722840 To: Cfadda <sip:225@82.187.90.68> Contact: "Cfadda" <sip:225@192.168.40.254:2660> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER Expires: 1800 Max-Forwards: 69 User-Agent: X-PRO release 1105x Content-Length: 0 Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) --- Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER request as basis request Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : 5060 (non-NAT) Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to 192.168.40.68:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.90.68>;tag=3189722840 To: Cfadda <sip:225@82.187.X.X > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:225@192.168.12.106> Content-Length: 0 --- Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to 192.168.40.68:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05f14a3e" Content-Length: 0 So asterisk sends response to the wrong interface. How can I solve this? Thanks in advance, C.

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[Serusers] DNS failover
by Nuno Ribeiro 05 Nov '08

05 Nov '08

Hi all, I have a scenario where I was testing DNS failover and this feature seems to fail. I will try to explain the scenario: I'm trying to establish a call between two user Agents, where one of the users has services associated (trigger to ASs) to the session establishment signaling (INVITE). So when SER makes the DNS query receives 2 ip addresses for this AS. The sip message will be correctly forwarded to one of this two entries and if in case of failure it will try to send the message to the next entry (second ip address). But when we are triggering to the AS we will add a route header and this header is not created in the second case, i.e. after DNS failover as the SIP INVITE will be created based in the SIP message received and not in the SIP message sent in the first place. So anyone has any thoughts about this question and a solution to it? Thanks in any advance. Best Regards -- Nuno Ribeiro

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[Kamailio-Users] SER/Asterisk interworking mailing list.
by Alex Balashov 05 Nov '08

05 Nov '08

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in duplicated effort and lack of specialised response. This is mainly due, I think, to the fact that detailed Asterisk experience - while common - is not a prerequisite for working with the SER products, while for Asterisk people SER can often be a next step in scalability and VoIP service delivery platform enhancement that they are just getting into. And so on. There's pollution in the respective discursive spaces; a lot of Asterisk people posting to the SER lists ask a lot of Asterisk-specific questions in addition to any they may have about SER which can be construed as potentially off-topic by some members, and the opposite is true on the Asterisk lists when detailed, involved discussion about SER occurs. We need to capture that discussion that exists at the overlap and is specifically concerned with making these two systems work together, requiring somewhat detailed and esoteric understanding of both and a community of user support and knowledge that focuses on both of these conceptual and product universes. Toward that end, I am hosting a new mailing list with this succinct purpose, if slightly unwieldy name, and encourage all interested to join. It is called 'SER-Asterisk-Interwork' and can be accessed for subscription here: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork The archives are available here: http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ You can post to the list at: ser-asterisk-interwork(a)lists.evaristesys.com It's the same GNU Mailman stuff you are already used to. While it could be argued that this cross-product discussion is valuable to retain in both communities, I think there is considerable benefit to creating a specialised mailing list that focuses specifically on this integration path and the unique interoperation and configuration issues it creates. I think it would be good to get some of this discussion off of the SER and Asterisk-specific mailing lists where it has somewhat marginal relevance at times and refocus it. If you agree and are interested in this topic, you are invited to join the list. One last note: The SER/OpenSER community has been in a state of flux recently, with OpenSER undergoing a name change to Kamailio and subsequently seeing a fork. The incumbent Kamailio project is now in the process of merging with the original SER project. The choice of nomenclature for list is not meant to imply an endorsement of or affinity for the IPTel SER project per se. It is just that right now it serves the aim of terseness to use a common denominator, to refer to this family of projects as the "SER ecosystem." Whether you are a SER, OpenSER, Kamailio, or OpenSIPS user, you are part of that "SER ecosystem." That is why the list is named what it is. Thank you, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Load userblacklist table to cache w/o restarting?
by Pablo Bertuzzi 05 Nov '08

05 Nov '08

Hello forum, I was wondering whether there is a fifo command or workaround to load the userblacklist table into memory without restarting. I have a large carrierroute table and it takes about 20 secs to start. best regards, PB

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Re: [Kamailio-Users] rewriteuser with AVP
by Samuel Muller 05 Nov '08

05 Nov '08

Hello Ravi , i'm using radiator but not freeradius, actually my users are stored into an user flat file. here you can have an example of user : 0123452000(a)sip.720.fr User-Password = "2000" User-Name = "0123452000", Sip-User-ID = "0123452000", Sip-User-Realm = "sip.720.fr", Sip-URI-User = "0123452000(a)sip.720.fr", Sip-Rpid = "0123452000", Sip-Group = "full", Sip-AVP = "asserted_id:0123452000" and concerning services attributes, you have to declare them like this : => in kamailio.cfg (or openser.cfg, ser.cfg, opensips.cfg) : modparam("acc|auth_radius", "service_type", 15) modparam("avp_radius", "caller_service_type", 30) modparam("avp_radius", "callee_service_type", 31) if you're using peering module : modparam("peering", "verify_destination_service_type", 21) modparam("peering", "verify_source_service_type", 22) => in the radius dictionary (for radiusclient-ng, and the dictionary used by your radius server) ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc/auth radius VALUE Service-Type Sip-Verify-Destination 20 # Proprietary, peering VALUE Service-Type Sip-Verify-Source 21 # Proprietary, peering VALUE Service-Type Sip-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type Sip-Callee-AVPs 31 # Proprietary, avp_radius => then, in your radius server config, you must add one handler per service-type example for radiator, just a base config based on flat user file: <Handler Service-Type=Sip-Verify-Destination> RewriteUsername s/^sip:(.*)/$1/ <AuthBy FILE> NoCheckPassword Filename %D/user.peer NoDefault </AuthBy> </Handler> I can't help you actually for a radius DB, 'cause i'm working on it. cheers, .Sam. -- Samuel MULLER Ingénieur Reseaux & Telecom 720 DEGRES +33 (0)663 128 505 sml(a)720.fr On Wed, Nov 5, 2008 at 1:28 PM, LetMeKnow < sunkara.raviprakash.feb14(a)gmail.com> wrote: > Hello Samuel, > Can you check the radius services attributes > > is you using users.conf or radius databases. > > can see this url > http://www.kamailio.org/docs/openser-radius-1.0.x.html#freeradius_users > > > > Thanks &Regards > Ravi Prakash Sunkara > VoIP Architect & JAVA-SIP Developer > +91-9999882776 > > > 2008/11/5 Samuel Muller <sml(a)720.fr> > >> Hello all, >> >> I've a little question : >> I got a username in an AVP by avp_load_radius, and I would rewrite the uri >> with it. >> >> The objectives is to replace the r-uri by an ipbx uri, then forward the >> call to it by a new branch (to not loose the phone number requested). >> in this case : 0123452000 is behind a spa9000. The user 0123452000 has an >> AVP called "ipbx", that i'm using to do some groups and permissions (plus >> the rewrite). >> >> actually, it does not work, i tried many ways : >> >> # load radius attributes of the callee >> if (!avp_load_radius("callee")) >> { >> xlog("L_INFO","-> user unknown in radius usr db : $ru"); >> route(15); # route PSTN >> } >> >> # verify the AVPs we got (caller and callee) >> xlog("L_INFO", " -AVP------------------------------------- "); >> avp_print(); >> xlog("L_INFO", " ----------------------------------------- "); >> >> # callee is behind an ipbx (avp ipbx) ? >> if (is_avp_set("$avp(s:callee_ipbx)")) >> { >> xlog("L_INFO", "-> callee is behind an ipbx : >> $avp(s:callee_ipbx)"); >> setflag(14); # flag IPBX >> #subst_user('/$rU$/$avp(s:callee_ipbx)/'); >> #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); >> rewriteuser($avp(s:callee_ipbx)); >> } >> >> # callee in usrloc ? >> if (lookup("location")) >> { >> append_hf("P-hint: usrloc applied\r\n"); >> xlog("L_INFO","-> registered user called : $tu"); >> route(7); # route FORWARD >> } >> else >> { >> xlog("L_INFO","-> 480: local user not found in usrloc : >> $tu"); >> sl_send_reply("480","Temporarily Unavailable"); >> drop; >> } >> >> and in the debugs log : >> >> Nov 5 13:04:51 ser0 kamailio[2930]: >> -ROUTE--INBOUND-------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth >> Success >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP >> 'callee_asserted_id'/0='0123452000'/0 has been added >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP >> 'callee_ipbx'/0='spa9000'/0 has been added >> Nov 5 13:04:51 ser0 kamailio[2930]: >> -AVP------------------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5cfe8, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<callee_ipbx> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<spa9000 / 7> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5cf88, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<callee_asserted_id> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<0123452000 / 10> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5ced8, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<caller_asserted_id> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<0123451011 / 10> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5ce38, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: >> ----------------------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( >> s(a)sip.720.fr' Not found in usrloc >> Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in >> usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: >> flags=ffffffffffffffff >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params >> 77.246.81.162, 192.168.0.134, 0 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying >> list 0x7fb3d7e5d040 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up >> >> >> -> But, if I did it by : >> >> if (uri =~ "^sip:012345200{1}") >> { >> rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> >> "); >> } >> if (lookup("location")) >> { >> append_hf("P-hint: usrloc applied\r\n"); >> xlog("L_INFO","-> registered user called : $tu"); >> route(7); # route FORWARD >> } >> >> it works great, and the call is ok (the ipbx forward correctly the request >> the phone behind him). >> >> >> Anyone has an idea ? thanks in advance ! >> >> >> -- >> Samuel MULLER >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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Re: [Kamailio-Users] Users Digest, Vol 42, Issue 11
by luzango mfupe 04 Nov '08

04 Nov '08

> 9. Re: the sip router project (Bogdan-Andrei Iancu) > > > Mmmh! guys, i am missing some points here, does this has anything to do > with OPENSIPS-YATE Collabo? a counter attack move of some sort??? > > > ------------------------------ > > Message: 9 > Date: Tue, 04 Nov 2008 17:24:15 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [Kamailio-Users] the sip router project > To: Johansson Olle E <oej(a)edvina.net> > Cc: devel <devel(a)lists.kamailio.org>, serusers(a)lists.iptel.org, > sr-dev(a)lists.sip-router.org, users <users(a)lists.kamailio.org>, SER > developer mailing list <serdev(a)iptel.org>, > business(a)lists.kamailio.org > Message-ID: <4910691F.9020509(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Olle, > > Thank you for your thoughts. On a first view, it looks interesting, but I'm > missing some points here (important points): > > 1) as OpenSER was forked from SER because different views (and the OpenSER > view proved to be a very popular and successful one), I wonder > why, Kamilio is getting back to SER? not sharing any more the OpenSER view > as claimed? because such merging will definitely have a great > impact on the dynamical and openness of the projects (like releases, > contributions, driving the project) > > 2) this major change of perspective (at least for kamilio) was a backstage > decision, kept secret from the community - shouldn't be in the > interest of the community to say if going back to the roots (merging into > SER) is something wanted or not? it somehow contradicts the self > existence of OpenSER, right? > > 3) the benefits you mentions are mainly optimization of the internal > project activities and not optimizations of the outcome - what the > project will deliver. And I guess this is the most important. We already > went though the experience of large devel community, frameworks, etc but > with no outcome for more than 2 years... > > >From my personal perspective, the new project looks more like SER > absorbing Kamilio (considering the sizes, the companies behind each > project, the resources, and the man-power behind each project). > > And at the moment I would like preserve the OpenSER vision and to have an > open source project (a standalone one), far away from the "control" of any > Big Brother ;) > > Regards, > Bogdan > > > > Johansson Olle E wrote: > > Congratulations to both projects! > > > > I think this is very beneificial for both communities as well as for > > all the companies out there using the two products. > > > > I am very happy with this result and hope that the third kid in the > > SER family - OpenSIPS - will consider joining the effort too. > > > > Best regards, > > /Olle > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 11 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] the sip router project
by Daniel-Constantin Mierla 04 Nov '08

04 Nov '08

Hello Everybody, We are pleased to announce to you the SIP Router project. It aims to build a solid open source SIP routing platform, based on collaboration of the SIP Express Router (SER) and Kamailio (OpenSER) teams. Developers of these two projects believe that an united and non-conflicting environment will bring many benefits to them, community members and companies: * bring together the developers and user communities of both projects * reduce maintenance overhead * avoid duplicated efforts in development * develop a core framework that is flexible, extensible and scalable * promote and build a solid open source SIP server project * ensure business credibility * make future forking undesirable, this harms everybody, affects credibility and business You are welcome to join! Visit the web site at: http://sip-router.org There is a meeting in Karlsruhe, Germany, on Nov 10, 2008, hosted by 1&1, where the developers and community members have the chance to discuss and tune the last aspects of the new project. We are looking to see many of you there: http://sip-router.org/index.php/meeting/ We hope this is a great news for you, thanks to the effort of main developers and management teams of the two projects. We invite you to join the new mailing list for further discussions: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev Daniel, Jiri Kamailio Management Team SER Management Team

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[Kamailio-Users] Unknown column 'from_uri' in 'field list'
by Kaeman Chris 04 Nov '08

04 Nov '08

Hello, modparam("acc", "db_extra", "from_uri=$fu; to_uri=$tu") added 2 new fields in the acc table. Logs giving following errors while making calls Nov 4 07:14:11 [4567] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'from_uri' in 'field list' Nov 4 07:14:11 [4567] ERROR:core:db_do_insert: error while submitting query Nov 4 07:14:11 [4567] ERROR:acc:acc_db_request: failed to insert into database Is this error due to the added fields? Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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[Kamailio-Users] SER config file adaptation to Kamailio ?
by Samuel Muller 04 Nov '08

04 Nov '08

Hello all, I tried during one week to adapt my old SER configuration file (ser.cfg, for SER 2.0.0), who works really great, to Kamailio v1.4.2. I used to change all the stuff needed (many functions, onsend route, ... disappeared or was replaced, ...). That's ok, concepts are the same - registration are ok, NAT (seems to) works with rtpproxy, and I'm using radius server (Radiator). MySQL is only here to store locations, domains, and trusted stuff like gateways. Everything looks fine (load kamalio : no pbm, REGISTER, OPTIONS methods and SIP ping ok, ...) Right now, RTP does not work, and no SIP phone (linksys spa9xx, thomson st2030, ...) can't call - there's no ringing, even if all the sip phones are registered and known by OpenSER ... It seems OpenSER can't send the SIP requests to the callee ??? Did anyone had the same problem and resolved it ? Or, eventually anyone does have some config sample config file (as you can have in SER examples stuff) ? I would like to let here a debug log of a call from a linksys spa942 to a sjphone but it's a bit enormous (debug log + config file) ... so i let it in attached file. the scenario is : 0123451011 (sjphone) calls 0123451012 (linksys spa9xx) the 2 UACs are on the same LAN, through the same router, with the same realm. there're registered and are stored into the location table in mySQL. but the problem is present in all the case we can have (different realms, different WAN connections, different SIP UACs ...) Thanks in advance ! -- Samuel MULLER

2 2

[Kamailio-Users] Remote-Party-ID header
by ALAEDDINE abbech 04 Nov '08

04 Nov '08

hi all, Please, what is the role of Remote-Party-ID header? thanks

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[Kamailio-Users] PUA API send to certain IP
by Schumann Sebastian 04 Nov '08

04 Nov '08

Anca, All, I have the following scenario UE - OpenSER Proxy/Presence Server - OpenSER Registrar The proxy acts as proxy and presence server. If the UE sends a subscribe to user@registrar, it handles this subscription. If it publishes its state user@registrar it does as well. All registration/call control is proxied to the registrar. I try now to use the PUA module API to create PUBLISH on the Proxy/Presence Server. This works fine, the only issue is that it has the request-uri user@registrar, like UE generated subscriptions. This is actually not an issue, but wanted. The problem is that the proxy forwards the package to the registrar acc. R-URI. I want it sent to another IP. Is there the possibility to define the IP where the package should be sent to, i.e. presence server IP, to handle the packet there? Thanks! Best regards, Sebastian

3 6

[Serusers] How to use SRV records
by Someone beneath the blue sky 04 Nov '08

04 Nov '08

Hello all, I'm pretty new to the SIP world and appreciate if somebody can help me out. In the SerWeb login page I noticed a link "Have-My-Domain!", and it looks like that the result would be having something similar to email forwarding service. Suppose that I have my own domain (e.g. mydomain.com), and I have added an SRV record to it and pointed it to iptel's sip server. Then following the link I was able to create a new username (e.g. user01). Now based on what I've understood, whenever somebody calls the sip uri user01(a)mydomain.com my soft phone that is registered with the above account should ring (my soft phone is using user01 for username, mydomain.com for server and iptel.org for registerar and it seems that it is registered without any problem). My questions: 1. What prevents others from impersonating a new username in my domain ? It seems that if somebody knows that mydomain.com has an SRV record pointed to iptel.org, can create another username (user02(a)mydomain.com) and use it without my knowledge, what is worse is that my DNS provider doesn't have any logging options for DNS requests, so I'll never find out about it. 2. In my account in iptel ( user01(a)mydomain.com ), there is another sip uri ( something like 123456(a)mydomain.com ), which I guess is supposed to be used by somebody who uses a device without alphabets (for example if somebody wants to call me via sipbroker gateways), I've already tried *478 - 123456 ( *478 is the code for iptel), but it doesn't work and it goes to voice mail, how can I use sipbroker to call user01(a)mydomain.com ? Thanks a lot in advance and sorry for the long message ! Someone

2 1

[Kamailio-Users] Proxy IAX ?
by BERGANZ François 04 Nov '08

04 Nov '08

Is there a proxy IAX ? Openser can do it ? Thank you Cordialement BERGANZ François

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[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

>> Hi all, >> i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: >> >> A -------------------------- SER ---------------------------------- B >> SIPp SIPp >> >> >> i have a problem using SER : >> I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. >> >> the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. >> >> i m putting below my config file ser.cfg >> >> i have also some questions: >> >> 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. >> 2/ also i have noticed that when using the command : serctl moni , i got the following message: >> root@amar-desktop:~# serctl moni >> [: 1074: ==: unexpected operator >> [: 1074: 0: unexpected operator >> >> despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. >> >> 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. >> >> i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. >> >> >> Amar >> >> >> >> >> _________________________________________________________________ >> You live life beyond your PC. So now Windows goes beyond your PC. >> http://clk.atdmt.com/MRT/go/115298556/direct/01/ > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

2 1

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

> > Hi all, > i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: > > A -------------------------- SER ---------------------------------- B > SIPp SIPp > > > i have a problem using SER : > I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. > > the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. > > i m putting below my config file ser.cfg > > i have also some questions: > > 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. > 2/ also i have noticed that when using the command : serctl moni , i got the following message: > root@amar-desktop:~# serctl moni > [: 1074: ==: unexpected operator > [: 1074: 0: unexpected operator > > despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. > > 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. > > i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. > > > Amar > > > > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

1 0

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

Hi all, i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: A -------------------------- SER ---------------------------------- B SIPp SIPp i have a problem using SER : I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. i m putting below my config file ser.cfg i have also some questions: 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. 2/ also i have noticed that when using the command : serctl moni , i got the following message: root@amar-desktop:~# serctl moni [: 1074: ==: unexpected operator [: 1074: 0: unexpected operator despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. Amar _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

1 0

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 03 Nov '08

03 Nov '08

Hello, being the time for a new IRC devel meeting, I propose Thursday, Nov 6, 2008, at 15:00UTC, on channel #kamailio, server irc.freenode.net The focus is to sync the developers related the work since the last major release, roadmap to next major release as well as minor releases scheduling. Please add to the agenda at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

Re: [Kamailio-Users] re mote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

________________________________ Monday, November 3, 2008 3:28:30 AM arif.zaman Wrote: > Did you check the location table whether callee is registered before > forwarding any call? that is strange that I see in the table location it, registered : +----+----------+------------------------------------------+------------------------+------------------------------+ | id | username | contact | received | user_agent | +----+----------+------------------------------------------+------------------------+------------------------------+ | 1 | 122 | sip:122@192.168.0.60:5065 | sip:190.184.35.4:5065 | Linksys/SPA942-5.2.8 | | 2 | 201 | sip:201@192.168.11.30:5060;transport=udp | sip:192.168.11.30:5060 | Grandstream BT200 1.1.4.18 | | 4 | 202 | sip:202@192.168.10.30:5064;transport=udp | sip:190.212.5.155:5064 | Grandstream GXP2020 1.1.6.16 | +----+----------+------------------------------------------+------------------------+------------------------------+ regardsss rickygm ________________________________

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[Kamailio-Users] INVITE accounting problem
by Kaeman Chris 03 Nov '08

03 Nov '08

Hello, I have a problem in accounting INVITE. When the call is connected with duration the INVITE, ACK, BYE are accounted in db. But the problem is when the call is disconnected before it is actually attempting to connect. i am getting only CANCEL accounted whereas INVITE is not accounted for the cancelled call. I have the xlogs showing INVITE and CANCEL both for this call only no accounting is performed for INVITE. following is the config for INVITE & CANCEL; if (is_method("INVITE")) { setflag(1); setflag(2); # MISSED if (allow_trusted()) { rewritehostport("xx.xxx.xy.xyz:5060"); route(1); } else { sl_send_reply("401", "Unauthorized"); route(1); exit; } } # CANCEL processing if (is_method("CANCEL")) { setflag(1); setflag(3); if (t_check_trans()) t_relay(); exit; } t_check_trans(); Please let me know what should be changed so that for the Call that is cancelled before it actually connects can have INVITE also accounted along w/ CANCEL. Thanks, KChris. Connect with friends all over the world. Get Yahoo! India Messenger at http://in.messenger.yahoo.com/?wm=n/

3 5

[Kamailio-Users] restrict calls from gateway to some extensions
by muhammad akl 03 Nov '08

03 Nov '08

Hey all , I want to make some ACL for the INVITES come from a PSTN-GW to certain users on my OpenSER server. for ex: I have the following extensions on my OpenSER server : 11001 11002 | | | | 11111 What I want to implement is something similar to this : Caller <-------> PSTN-GW <------->OpenSER <----> (complete this call to this extension ) 11001 Caller <-------> PSTN-GW <------->OpenSER <----> (reject this call for this user ) 11002 and so on for other extensions . any way to implement this feature in openser ?

2 2

[Kamailio-Users] remote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

Hello list, I have another question and a very strange detail, I have a couple of connected remote clients to my kamailio 1.3.2 and mediaproxy 1.9.1, when the client 122 calls to the client 202 this very contacts or, if I call a minute I contact him very well, but if I call to the client 122 after some minutes I don't contact with that client and the jumps me to the voicemail . I see in the chart location the address ip of the UAC that it can be happening? regards rickygm 122 UAC <-> NAT <-> internet==== kamailio 1.3.2 / mediaproxy ==== internet <-> ADSL <-> NAT <-> UAC 202 public ip: x.x.x.x.x interface: any filter: (ip) and ( port 5060 ) # U +0.890415 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000286 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE Proxy-Authenticate: Digest realm="190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.197976 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.073962 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003823 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9;rport=5064;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000053 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.485936 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.999801 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +1.999612 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999219 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999220 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +4.001208 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.599535 190.184.22.152:5060 -> 192.168.11.1:5070 INVITE sip:u122@192.168.11.1:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000362 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Length: 0 # U +0.000283 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 15334 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.002152 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 P-hint: onreply_route|usemediaproxy v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.236892 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK0fd7eaebf7c6f6c6 Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000248 190.184.22.152:5060 -> 190.184.22.152:5070 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.3 Via: SIP/2.0/UDP 192.168.10.30:5064;received=190.212.35.223;branch=z9hG4bK0fd7eaebf7c6f6c6 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 exit 40 received, 0 dropped

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 02 Nov '08

02 Nov '08

Hi list, always looking for as solving my audio problem with mediaproxy asterisk and openser, there will be some form of telling to the openser that when he comes from the from sip:asterisk@192.168.10.1:5070 that doesn't use the mediaproxy or the onreply_route[1] , SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK36b7f619;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as42edbc9b;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as42edbc9b To: <sip:113@192.168.10.1>;tag=6d45d2188218c8ef Call-ID: 5a1c60382f18bd832fa3bdc54dc6ab13(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35064 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 making other tests if I change this form the onreply_route[1], I have audio in the openser extension, but the one that this behind the pstn doesn't have audio or he doesn't listen to me onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } my best regardss rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

________________________________ Friday, October 31, 2008 2:59:58 AMDaniel-Constantin Mierla wrote: > do you have to bridge the RTP between the two network interfaces? >I do not know mediaproxy, but rtpproxy (nathelper module) has bridge mode >that can be used in such scenarios. Hi Daniel, I don't have a bridge with the two net cards, I had read some rtpproxy that can be used in bridge mode, but I didn't find alone examples some comments, but he gave me an error and I could not make it work this problem has me sad... and I am not seeing an exit.. regards .. rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Thank you Luciano for your answer but doesn't it still work me, do the calls enter me of the pstn and don't I listen anything... where I can look for? , there will be some of form so that the IP of where the calls come from asterisk does not use mediaproxy, somebody has an example. any help is appreciated.. thank you rickygm ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Sent: Friday, October 31, 2008 5:19:25 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); USE MEDIA PROXY HERE---> use_media_proxy(); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com> wrote: I show you here my openser.cfg my best regards rickygm loadmodule "nathelper.so" loadmodule "mediaproxy.so" modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("nathelper","rtpproxy_sock", "") modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!is_method("REGISTER")) { if(nat_uac_test("19")){ record_route(";nat=yes"); } else { record_route(); }; }; if (has_totag()) { # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; if (method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } ##Loose_route packets if (loose_route()) { if(!has_totag()){ sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; }; route(1); }; #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); if (method=="REGISTER"){ route(2); } else { route(3); }; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { ; if (method=="INVITE" || method=="ACK") { ; end_media_session(); ; }; # sl_reply_error(); }; exit; } route[2] { # # -- Register request handler -- # if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); fix_nated_register(); force_rport(); }; save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { # # -- INVITE request handler -- # if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if (client_nat_test("3")) { append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); setbflag(7); force_rport(); fix_contact(); }; if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(4); route(1); exit; } consume_credentials(); # Verify aliases lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound route(10); } else { # -- Inbound to outbound route(11); }; } else { # From an external domain -> do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { #-- Outbound to inbound route(12); } else { # -- Outbound to outbound route(13); }; }; } route[6] { # # -- NAT handling -- # if (isbflagset(6) || isbflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:[2346578][0-9]{6}@.*") { if (is_user_in("credentials", "local")){ # rewritehostport("192.168.10.1:5070"); route(6); route(4); exit; } else { sl_send_reply("403", "No tienes permiso para llamadas locales"); exit; }; }; if (!lookup("location")) { if (does_uri_exist()) { revert_uri(); prefix("u"); rewritehostport("192.168.10.1:5070"); route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; }; sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } route[4] { rewritehostport("192.168.10.1:5070"); route(1); } ### aqui se detecta cunado una llamada es no disponible , ocupada o falla failure_route[1] { if (t_was_cancelled()) { exit; }; if (t_check_status("486")) { revert_uri(); prefix("b"); xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; } onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Serusers] SER/FIFO not accessible
by Shruti Sanadhya 31 Oct '08

31 Oct '08

Hi all, I am trying to setup a SER to support calls on Kphone. I am trying to add users to the mysql SER database using: serctl 1000 1000 1000(a)mydomain.com but I am getting the 'SER/FIFO not accessible' error. I tried it using both the SER and root password for MySql, but no luck. I have also added 'fifo_mode=0777" in the configuration file, but the error still persists. Please advise, Thanks!!

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by luzango mfupe 31 Oct '08

31 Oct '08

Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); *USE MEDIA PROXY HERE---> use_media_proxy(); *}; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com>wrote: > > I show you here my openser.cfg > > my best regards > > rickygm > > loadmodule "nathelper.so" > loadmodule "mediaproxy.so" > > > modparam("registrar", "received_avp", "$avp(i:42)") > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "nat_bflag", 4) > > modparam("nathelper","rtpproxy_sock", "") > > modparam("nathelper","natping_interval", 0) > modparam("nathelper","received_avp", "$avp(i:42)") > modparam("mediaproxy","natping_interval",20) > modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") > modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > > route{ > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!is_method("REGISTER")) { > if(nat_uac_test("19")){ > record_route(";nat=yes"); > } else { > record_route(); > }; > }; > > if (has_totag()) { > # subsequent messages withing a dialog should take the > # path determined by > record-routing > if (loose_route()) { > if(method=="BYE") { > #Account BYE transactions > setflag(2); > }; > if (method=="INVITE" && (!allow_trusted())) { > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Forbidden, use From=ID"); > exit; > > }; > > }; > if(method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > ##Detect requests in the dialog behind NAT and flag with 6 > if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ > append_hf("P-hint: LR|fixcontact,setflag6\r\n"); > fix_contact(); > setbflag(6); > }; > route(1); > } else { > sl_send_reply("404", "Not here"); > } > exit; > } > > ##Loose_route packets > if > (loose_route()) { > if(!has_totag()){ > sl_send_reply("403", "Initial Loose-Routing Rejected"); > exit; > }; > > > route(1); > }; > > > #CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > end_media_session(); > t_relay(); > }; > exit; > } > t_check_trans(); > > if (method=="REGISTER"){ > route(2); > } else { > route(3); > }; > } > > route[1] { > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > t_on_reply("1"); > t_on_failure("1"); > if (!t_relay()) { > ; if (method=="INVITE" || method=="ACK") { > ; end_media_session(); > ; }; > > > # sl_reply_error(); > }; > exit; > } > route[2] { > # > # -- Register request handler -- > # > if (is_uri_host_local()) { > if (!www_authorize("", "subscriber")) { > > www_challenge("", "0"); > exit; > }; > if (!check_to()) { > sl_send_reply("403", "Forbidden"); > exit; > }; > sl_send_reply("100", "Trying"); > if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setbflag(6); > fix_nated_register(); > force_rport(); > }; > > save("location"); > exit; > } else if { > sl_send_reply("403", "Forbidden"); > }; > } > > route[3] { > # > # -- INVITE request handler -- > # > if (method=="INVITE") { > # Account INVITE packets > setflag(2); > # Account Missed calls > setflag(3); > }; > if (is_from_local()){ > # From an internal domain -> check the credentials and the FROM > if(!allow_trusted()){ > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Forbidden, use From=ID"); > exit; > > }; > > }; > if (client_nat_test("3")) { > append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); > setbflag(7); > force_rport(); > fix_contact(); > }; > > if(avp_db_load("$ru/username","$avp(s:callfwd)")) { > avp_pushto("$ru", "$avp(s:callfwd)"); > route(4); > route(1); > > exit; > } > > consume_credentials(); > > # Verify aliases > lookup("aliases"); > if (is_uri_host_local()) { > # -- Inbound to Inbound > > route(10); > } else { > # -- Inbound to outbound > route(11); > }; > > } else { > # From an external domain -> do not check credentials > #Verify aliases, if found replace R-URI. > lookup("aliases"); > if (is_uri_host_local()) { > #-- Outbound to inbound > route(12); > } else { > # -- Outbound to outbound > route(13); > }; > }; > } > route[6] { > > # > # -- NAT handling -- > # > if (isbflagset(6) || isbflagset(7)) { > append_hf("P-hint: Route[6]: mediaproxy \r\n"); > use_media_proxy(); > }; > } > > route[10] { > #from an internal domain -> inbound > #Native SIP destinations are handled using the location table > append_hf("P-hint: inbound->inbound \r\n"); > if (uri=~"^sip:[2346578][0-9]{6}@.*") { > if (is_user_in("credentials", "local")){ > # rewritehostport("192.168.10.1:5070"); > route(6); > route(4); > exit; > } else { > sl_send_reply("403", "No tienes permiso para llamadas locales"); > exit; > }; > }; > > if (!lookup("location")) { > if (does_uri_exist()) { > revert_uri(); > prefix("u"); > rewritehostport("192.168.10.1:5070"); > route(6); > route(1); > } else { > sl_send_reply("404", "Not Found"); > exit; > }; > sl_send_reply("404", "Not Found"); > exit; > }; > > > route(6); > route(1); > } > > route[11] { > # from an internal domain -> outbound > # Simply route the call outbound using DNS search > append_hf("P-hint: inbound->outbound \r\n"); > route(1); > } > route[12] { > # From an external domain -> inbound > # Verify aliases, if found replace R-URI. > lookup("aliases"); > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > exit; > }; > route(1); > } > route[13] { > #From an external domain outbound > #we are not accepting these calls > append_hf("P-hint: outbound->inbound \r\n"); > sl_send_reply("403", "Forbidden"); > exit; > } > > > route[4] { > rewritehostport("192.168.10.1:5070"); > route(1); > > } > > ### aqui se detecta cunado una llamada es no disponible , ocupada o falla > failure_route[1] { > if (t_was_cancelled()) { > exit; > }; > > if (t_check_status("486")) { > revert_uri(); > > prefix("b"); > xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); > rewritehostport("192.168.10.1:5070"); > append_branch(); > route(1); > exit; > }; > if (t_check_status("408") || t_check_status("480")) { > revert_uri(); > prefix("u"); > xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); > rewritehostport("192.168.10.1:5070"); > append_branch(); > route(1); > exit; > }; > } > onreply_route[1] { > # > #-- On-replay block routing -- > # > if (client_nat_test("1")) { > append_hf("P-hint: Onreply-route - fixcontact \r\n"); > fix_nated_contact(); > }; > > if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { > if (search("^Content-Type:[ ]*application/sdp")) { > append_hf("P-hint: onreply_route|usemediaproxy \r\n"); > use_media_proxy(); > }; > }; > > exit; > } > > > > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, I have gotten myself an ip it public with my supplier, I have two network cards in the pc that I am using for openser and mediaproxy together with asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the pstn through asterisk I don't have audio, if I call to the pstn they listen to me well . From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: path Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7492 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 INVITE sip:2685249@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:2685249@192.168.10.1", nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", response="c40a9387fdf5de29115c1edadc7f79db" Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 358 v=0 o=112 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03;rport=5060 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: replaces, timer, path Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 358 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=112 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2685249@192.168.10.1:5070> Content-Length: 0 I don't have a lot of experience with mediaproxy, and I have some doubts that such you see they can help me to clarify, inside the file mediaproxy.ini some options appear which I have configured them but I am not sure if it is the best way. my scenario is the following one: <-> UAC<-> NAT <-> ADSL <-> Internet <-> eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (192.168.11.1) <-> UAC [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = openser listen = None allow = None proxyIP = x.x.x.x (public ip) ;portRange = 60000:65000 portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 [Accounting] ; one of none, radius or database accounting = none [Database] user = dbuser password = dbpass host = dbhost database = radius table = radacct [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary retries = 2 timeout = 3 this couple of you line inside the openser, I don't still understand them according to the guide of ser getting started they are for asymmetric clients, but I don't find an example modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") somebody that can give me a good help... regards rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

I don't have problems when I make calls to the pstn I listen well and people listen to me well, the problem is when I receive a call from the pstn I don't listen anything and they don't listen to me, inside the sip.conf already has configured the values nat, externip localnet . I believe that the problem is that openser detects as nat the ip of my asterisk, eye > "I have the openser and the mediaproxy with asterisk in the same pc" ### Sip Log Asterisk #### <--- SIP read from 192.168.10.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact <-------------> --- (12 headers 0 lines) --- -- SIP/openser-08c0ea58 is ringing xserver*CLI> <--- SIP read from 192.168.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.64:35004 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.64:35004 list_route: hop: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> set_destination: Parsing <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> for address/port to send to set_destination: set destination to 192.168.10.1, port 5060 Transmitting (NAT) to 192.168.10.1:5060: ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK32b6019c;rport Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/openser-08c0ea58 answered Zap/4-1 xserver*CLI> <--- SIP read from 192.168.10.1:5060 ---> BYE sip:asterisk@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0 Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0 From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f Supported: path Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 9793 BYE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.10.1 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0 Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d> From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 9793 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:asterisk@192.168.10.1:5070> Content-Length: 0 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Hi RickyI should have seen how you handle NAT in kamaiilo.conf but you can also edit sip.conf in Asterisk and try to put Nat=yes Rgds,

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[Kamailio-Users] Fw: version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 30 Oct '08

30 Oct '08

OK thank you Nuno to answer, but I continue thinking that that version is not compatible with mediaproxy 1.9.1, for comments that read by the web, but better we ask to ag-projects Dan .. my best regards rickygm ________________________________ From: Nuno Marques <yangsengopenser(a)gmail.com> To: Ricky Gutierrez <xserverlinux(a)yahoo.com> Cc: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:24:35 PM Subject: Re: version cdrtool mediaproxy 1.9.1 Hi Ricky, I'm using CDRTool 6.6.10 (latest). I still can't establish connection between CDRTool and mediaproxy database... but i'm almost there ! :) Nuno 2008/10/29 Ricky Gutierrez <xserverlinux(a)yahoo.com> Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

1 0

[Kamailio-Users] Dimensioning a telephony system based on openser!
by Nuno Marques 30 Oct '08

30 Oct '08

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites: - 10000 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity; Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance, Nuno

8 19

Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 30 Oct '08

30 Oct '08

Hi Henning Westerholt, Thanks for you reply... But I would like to know if there is a number that improve the performance... For example: If come 100 INVITE at the same moment, is there a NUMBER OF CHILDREN that has the best perfomance? The number of children depends on something? Thanks --- Em qua, 29/10/08, Henning Westerholt <henning.westerholt(a)1und1.de> escreveu: De: Henning Westerholt <henning.westerholt(a)1und1.de> Assunto: Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process Para: users(a)lists.kamailio.org Cc: dcardosoa(a)yahoo.com.br Data: Quarta-feira, 29 de Outubro de 2008, 11:25 On Wednesday 29 October 2008, Daniel-Constantin Mierla wrote: > > Is there a number of children that the Kamailio/Openser for improve > > the performance? > > > > Each process takes care of a request? > > yes, each process takes care of a SIP message (request/reply). Hi Daniel Cardoso, take a look to the core cookbook in our wiki [1] for more informations how you can control the number of childrens per protocol (UDP, TCP..). Cheers, Henning [1] http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#children Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

4 3

[OpenSER-Users] Openser1.3.2 + snmp
by Ricardo Alves Ferreira Silva 30 Oct '08

30 Oct '08

Hi all, Trying to install Openser1.3.2.... everything is OK and fine... but when I try to add support for snmp there are errors.. Dig dep into google, recheck all dependencies and didn't succeed... My env is FreeBSD 7.0-STABLE i386 The full net-snmp package is installed: atila# pkg_info | grep net-snmp net-snmp-5.3.2_3 An extendable SNMP implementation These are the error messages.... make modules/libsnmpstats modules.... Compiling alarm_checks.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -DMOD_NAME='"snmpstats"' -DNAME='"openser"' -DVERSION='"1.3.2-notls"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 4.2.1"' -D__CPU_i386 -D__OS_freebsd -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_NETINET_IN_SYSTM -DHAVE_KQUEUE -DHAVE_SELECT -c alarm_checks.c -o alarm_checks.o In file included from alarm_checks.c:38: openserObjects.h:41:38: error: net-snmp/net-snmp-config.h: No such file or directory openserObjects.h:42:40: error: net-snmp/net-snmp-includes.h: No such file or directory openserObjects.h:43:52: error: net-snmp/agent/net-snmp-agent-includes.h: No such file or directory If I edit each file and hardcode the path ... /usr/local/include/net-snmp/... there are no errors... What am I missing in making sure the path to the include files are OK... Tried a couple options but didn't succeed... tehre are tons of include files into the src.... Regards Ricardo Ferreira Yes, we do run Openser www.vipway.com.br Enterprise VoIP Services

5 10

[Serusers] Error in jabber module - XJAB:extract_aor: Error while parsing URI
by Tomasz Zieleniewski 30 Oct '08

30 Oct '08

Hi, We are using ser 2.0 We encountered the following error: 7(407) route[PRESENCE_SUBSCRIBE](SUBSCRIBE cseq:933): (handle_subscription(jabber)) 7(407) XJAB:extract_aor: Error while parsing URI But still SER sends an 200 OK response and after then a NOTIFY. Here is the message received by ser: SUBSCRIBE sip:lkl%touk.pl@jabber.rd.touk.pl SIP/2.0 Via: SIP/2.0/UDP 192.168.0.165;rport;branch=z9hG4bKxakietyi Max-Forwards: 70 Proxy-Authorization: Digest username="tzl",realm="touk.pl",nonce="490987cb2702bb9db8d631e016825ad1c9cd12ac",uri="sip:lkl%touk.pl@jabber.rd.touk.pl",response="bd4474cf8af3d1ec8a7246872c0ecb46",algorithm=MD5,cnonce="mk57n39293",qop=auth,nc=00000001 To: <sip:lkl%touk.pl@jabber.rd.touk.pl> From: "Tomasz Zieleniewski" <sip:tzl@sip.rd.touk.pl>;tag=cejrf Call-ID: vxiulpsynbnidjo(a)sen.rd.touk.pl CSeq: 933 SUBSCRIBE Contact: <sip:tzl@192.168.0.165> Accept: application/pidf+xml Event: presence Expires: 3600 User-Agent: Twinkle/1.2 Content-Length: 0 Jabber config: # -- jabber params --- modparam("jabber", "jaddress", "touk.pl") modparam("jabber", "jport", 5222) modparam("jabber", "registrar", "serimgw(a)sip.rd.touk.pl") modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.rd.touk.pl=%") Please give us a hint what could be wrong. Kind regards Tomasz

1 0

[Serusers] Configuring SER as an outbound proxy for multiple Ekigas on a LAN
by Enrico Maria Crisostomo 30 Oct '08

30 Oct '08

Hi all. I'm a first-time would-be user of SER and I have the following problem: I was planning to use Ekiga on a LAN with several workstations and I was looking for documentation about the options I have and the best setup I can achieve. How to solve the problem of NAT traversal for multiple Ekigas was answered by Ekiga documentation: use a SIP proxy. The workstations are all equipped with Solaris 10 and I have a small Solaris 10 server too, so I thought it was natural for me to use SER as a SIP proxy. Unfortunately I'm finding no documentation about how to configure it as a simple outbound proxy and to say the truth I found very little documentation about configuring a SIP proxy in general. My idea would be just having SER behave as an outbound SIP proxy and use ekiga.net accounts for everybody (no need for a PBX as Asterisk). It seems a kind of basic configuration but I have no idea where to start from. Do you have suggestions and pointers to documentation to help me achieve this goal? Thank you in advance, Enrico. -- Ελευθερία ή θάνατος GPG key: 1024D/FD2229AF fpr: 9E07 D40E 33A5 5993 6FC5 09A8 5BCF B1F2 FD22 29AF

2 1

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 30 Oct '08

30 Oct '08

Hi luzano thank you for your help and time, this it is my full ngrep. I have asterisk, mediaproxy and openser together in the same pc I have the doubt that when a incoming call from the PSTN by asterisk, sends it to an extension of openser, to which I do not request authentication to him within invites, I believe that there it is where I have problems with mediaproxy that it leaves you want to see of the openser.cfg, or everything? regards .. interface: any filter: (ip) and ( port 5060 ) # U +0.063740 192.168.10.1:5070 -> 192.168.10.1:5060 INVITE sip:113@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 14750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062 INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Content-Length: 0 # U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.000014 192.168.1.64:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.956449 192.168.10.19:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000027 192.168.1.64:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.156145 190.184.35.4:5060 -> 192.168.1.64:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000557 192.168.10.1:5070 -> 192.168.10.1:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 # U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 P-hint: LR|fixcontact,setflag6 # U +0.886545 192.168.10.28:5060 -> 192.168.10.1:5060 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> To: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:12:29 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango.

2 1

[Kamailio-Users] kamailio and rtpproxy-no audio
by Fabian Borot 30 Oct '08

30 Oct '08

hello I am trying to use rtpproxy and kamailio. Below please find my kamailio.cfg file. But this is what is happening: 1- ext 101[IP 76.109.183.2] is able to send audio to kamailio's IP [65.111.185.187]but it does not reach ext 100 IP [76.109.15.75]. 2- ext 100 does not send RTP at all. 3- I still see the private IPs in the SDPs of both exts. I am aattaching a wireshark capture Thank you in advance Fabian # $Id: kamailio.cfg 4576 2008-08-05 16:22:38Z klaus_darilion $ # # Kamailion basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = '/usr/local/etc/kamailio/tls/user/user-cert.pem' #tls_private_key = '/usr/local/etc/kamailio/tls/user/user-privkey.pem' #tls_ca_list = '/usr/local/etc/kamailio/tls/user/user-calist.pem' port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:65.111.185.187:5060 ####### Modules Section ######## #set module path mpath='/usr/local/lib/kamailio/modules/' /* uncomment next line for MySQL DB support */ loadmodule 'db_mysql.so' loadmodule 'sl.so' loadmodule 'tm.so' loadmodule 'rr.so' loadmodule 'maxfwd.so' loadmodule 'usrloc.so' loadmodule 'registrar.so' loadmodule 'textops.so' loadmodule 'mi_fifo.so' loadmodule 'uri_db.so' loadmodule 'uri.so' loadmodule 'xlog.so' loadmodule 'acc.so' /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule 'auth.so' loadmodule 'auth_db.so' /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'alias_db.so' /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see 'multi-module params' section ) */ #loadmodule 'domain.so' /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'presence.so' #loadmodule 'presence_xml.so' loadmodule 'nathelper.so' # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam('mi_fifo', 'fifo_name', '/tmp/kamailio_fifo') # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam('rr', 'enable_full_lr', 1) # do not append from tag to the RR (no need for this script) modparam('rr', 'append_fromtag', 0) # ----- rr params ----- modparam('registrar', 'method_filtering', 1) /* uncomment the next line to disable parallel forking via location */ # modparam('registrar', 'append_branches', 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam('registrar', 'max_contacts', 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam('uri_db', 'use_uri_table', 0) modparam('uri_db', 'db_url', '') # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam('acc', 'early_media', 1) modparam('acc', 'report_ack', 1) modparam('acc', 'report_cancels', 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable 'append_fromtag' in 'rr' module */ modparam('acc', 'detect_direction', 0) /* account triggers (flags) */ modparam('acc', 'failed_transaction_flag', 3) modparam('acc', 'log_flag', 1) modparam('acc', 'log_missed_flag', 2) /* uncomment the following lines to enable DB accounting also */ modparam('acc', 'db_flag', 1) modparam('acc', 'db_missed_flag', 2) # ----- usrloc params ----- modparam('usrloc', 'db_mode', 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam('usrloc', 'db_mode', 2) #modparam('usrloc', 'db_url', # 'mysql://openser:openserrw@localhost/openser') modparam('usrloc', 'nat_bflag', 3) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam('auth_db', 'calculate_ha1', yes) modparam('auth_db', 'password_column', 'password') modparam('auth_db', 'db_url', 'mysql://openser:openserrw@localhost/openser') modparam('auth_db', 'load_credentials', '') # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam('alias_db', 'db_url', # 'mysql://openser:openserrw@localhost/openser') # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam('domain', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('domain', 'db_mode', 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam('alias_db|auth_db|usrloc|uri_db', 'use_domain', 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam('presence|presence_xml', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('presence_xml', 'force_active', 1) #modparam('presence', 'server_address', 'sip:192.168.1.2:5060') # ----- nathelper params ----- modparam('nathelper', 'natping_interval', 10) modparam('nathelper', 'ping_nated_only', 1) modparam('nathelper', 'rtpproxy_sock', 'udp:localhost:8899') modparam('nathelper', 'rtpproxy_disable_tout', 20) modparam('nathelper', 'rtpproxy_tout', 2) modparam('nathelper', 'rtpproxy_retr', 2) modparam('nathelper', 'sipping_from', 'sip:pinger@mykamailio.com') modparam('nathelper', 'sipping_bflag', 7) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header('10')) { sl_send_reply('483','Too Many Hops'); exit; } if (is_method('REGISTER')) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize('', 'subscriber')) { www_challenge('', '0'); exit; } if (!save('location')) sl_reply_error(); exit; } if (!lookup('location')) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply('404', 'Not Found'); exit; case -2: sl_send_reply('405', 'Method Not Allowed'); exit; } } # when routing via usrloc, log the missed calls also setflag(2); # record routing if (!is_method('REGISTER|MESSAGE')) { record_route(); } # account only INVITEs if (is_method('INVITE')) { setflag(1); # do accounting } # for INVITEs enable some additional helper routes if (is_method('INVITE')) { force_rtp_proxy(); } if (!t_relay()) { sl_reply_error(); }; exit; # CANCEL processing if (is_method('CANCEL|BYE')) { unforce_rtp_proxy(); if (t_check_trans()) t_relay(); exit; } } _________________________________________________________________ Stay organized with simple drag and drop from Windows Live Hotmail. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_102008

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[Kamailio-Users] version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 29 Oct '08

29 Oct '08

Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

Hi I download Twinkle, and install all the prerequisites, except for Qt. I have some problems with it. First I download Qt for Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't know if it is the one I have to download, because In the Twinkle page, it only says you have to install "Qt for open source (3.3.0 or later, but not Qt4)". Then I do all the steps cd /tmp gunzip qt-x11-opensource-src-4.4.3.tar.gz # uncompress the archive tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it cd /tmp/qt-x11-opensource-src-4.4.3 ./configure #It took more than 6 hours make su -c "make install" Then I retry to install Twinkle entering in Twinkle directory an doing ./configure but I receive the following error: checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. So I read the FAQ of Twinke (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that I can skip this error by doing ./configure --disable-qt-check, but now I receive the following error: checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. So I realize that I didn't have libqt-mt so I try to recompile QT with "./configure -thread" but it didn't let me, so I try with "./configure -DQT_THREAD_SUPPORT" and the compilation took also mother tan 6 hours, and I retry to install Twinkle, but without success, I receive the same message. checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. I don't Know what I am doing wrong. Any Help!!!!! please. David Cespedes. ----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Martes, Octubre 28, 2008 8:15 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > is twinkle able now to display rich presence status? Last time I > tried> was showing only online/offline status. > > For now just online/offline status. > > But it has other great features as: > - Indication of Message Composition for Instant Messaging - RFC 3994 > - PingPong TCP keepalive. > - TCP/UDP > - PRACK > - q-value in registration. > > For me it's the most reliable softphone. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

Excuse me list, it was an error copy to this list, but maybe some one can help me. ----- Mensaje original ----- De: Alex Balashov <abalashov(a)evaristesys.com> Fecha: Miércoles, Octubre 29, 2008 11:31 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > This is not the Twinkle list. > > ingdavidcespedes(a)cable.net.co wrote: > > > Hi > > > > I download Twinkle, and install all the prerequisites, except for > Qt. I have some problems with it. First I download Qt for > Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't > know if it is the one I have to download, because In the Twinkle > page, it only says you have to install "Qt for open source (3.3.0 > or later, but not Qt4)". Then I do all the steps > > > > cd /tmp > > gunzip qt-x11-opensource-src-4.4.3.tar.gz # > uncompress the archive > > tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it > > cd /tmp/qt-x11-opensource-src-4.4.3 > > ./configure #It took more than 6 > hours> make > > su -c "make install" > > Then I retry to install Twinkle entering in Twinkle directory an > doing ./configure but I receive the following error: > > > > checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not > found in the pkg-config search path. Perhaps you should add the > directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment > variable No package 'qt-mt' found Package qt-mt was not found in > the pkg-config search path. Perhaps you should add the directory > containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable > No package 'qt-mt' found > > configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < > 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment > variable if your libraries are in a nonstandard prefix so pkg- > config can find them. > > > > So I read the FAQ of Twinke > (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that > I can skip this error by doing ./configure --disable-qt-check, but > now I receive the following error: > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > So I realize that I didn't have libqt-mt so I try to recompile QT > with "./configure -thread" but it didn't let me, so I try with > "./configure -DQT_THREAD_SUPPORT" and the compilation took also > mother tan 6 hours, and I retry to install Twinkle, but without > success, I receive the same message. > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > I don't Know what I am doing wrong. Any Help!!!!! please. > > > > David Cespedes. > > > > ----- Mensaje original ----- > > De: Iñaki Baz Castillo <ibc(a)aliax.net> > > Fecha: Martes, Octubre 28, 2008 8:15 am > > Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > > > >> 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >>> is twinkle able now to display rich presence status? Last time > I > >> tried> was showing only online/offline status. > >> > >> For now just online/offline status. > >> > >> But it has other great features as: > >> - Indication of Message Composition for Instant Messaging - RFC > 3994>> - PingPong TCP keepalive. > >> - TCP/UDP > >> - PRACK > >> - q-value in registration. > >> > >> For me it's the most reliable softphone. > >> > >> > >> -- > >> Iñaki Baz Castillo > >> <ibc(a)aliax.net> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Fwd: How to enable local_route feature ?
by Pascal Maugeri 29 Oct '08

29 Oct '08

Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal ---------- Forwarded message ---------- From: Pascal Maugeri <pascal.maugeri(a)gmail.com> Date: Wed, Oct 29, 2008 at 11:17 AM Subject: How to enable local_route feature ? To: Henning Westerholt <henning.westerholt(a)1und1.de> Cc: devel(a)lists.openser.org, openser users <users(a)lists.openser.org> Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal On Thu, Jun 26, 2008 at 1:11 PM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > On Friday 06 June 2008, Bogdan-Andrei Iancu wrote: > > OpenSER has a new type of route: local route. This route is to be > > executed when a brand new request is internally generated (openser being > > the UAC) ; such requests are result of : > > - t_uac_dlg MI command > > - msilo, sms, presence, cpl-c, dialog (and others) modules > > [..] > > Hello all, > > the addition of this feature short before the code freeze caused some > conflicts, which could not resolved sufficently in time for this release. > Thus the OpenSER board was asked to decide in this matter. The majority of > the board voted for decativating this feature in the 1.4 release series. > > For more details please take a look at the developer mailing list at [1], > [2] > and [3]. Details about the "dectivate" commit can be found at [4] > > The local route functionality is still available, but its not activated in > the > default installation. In order to use this feature its necessary to set > the "USE_LOCAL_ROUTE" define in the "Makefile.defs" file and compile your > own > binary from the source. Then it will possible to use this feature. > > In the 1.5.0 release this functionality (or something resemble this) will > be > present, fully usable and supported. > > With best regards, > > Henning Westerholt > > > [1] http://lists.openser.org/pipermail/devel/2008-June/014028.html > [2] http://lists.openser.org/pipermail/devel/2008-June/014122.html > [3] http://lists.openser.org/pipermail/devel/2008-June/014204.html > [4] http://lists.openser.org/pipermail/devel/2008-June/014444.html > > > -- > Henning Westerholt - Development Consumer Products / DSL Core > 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 29 Oct '08

29 Oct '08

Hi people... Is there a number of children that the Kamailio/Openser for improve the performance? Each process takes care of a request? Thanks in advance. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by luzango mfupe 29 Oct '08

29 Oct '08

Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango. > > Message: 2 > Date: Tue, 28 Oct 2008 23:10:12 -0700 (PDT) > From: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Subject: [Kamailio-Users] I don't have asterisk audio to openser - > mediaproxy > To: users(a)lists.kamailio.org > Message-ID: <654712.75990.qm(a)web59903.mail.ac4.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, > I have gotten myself an ip it public with my supplier, I have two network > cards in the pc that I am using for openser and mediaproxy together with > asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the > pstn through asterisk I don't have audio, if I call to the pstn they listen > to me well . > > > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7492 ACK > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Length: 0 > > > # > U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 > INVITE sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Proxy-Authorization: Digest username="112", realm="192.168.10.1", > algorithm=MD5, uri="sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>", > nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", > response="c40a9387fdf5de29115c1edadc7f79db" > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.10.30 > t=0 0 > m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;branch=z9hG4bKf428b928c25dad03;rport=5060 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > Server: OpenSER (1.3.2-notls (i386/linux)) > Content-Length: 0 > > > # > U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 > INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 69 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > P-hint: route(3)|setflag7,forcerport,fix_contact > P-hint: inbound->inbound > P-hint: Route[6]: mediaproxy > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.1.64 > t=0 0 > m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received= > 192.168.10.1 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:2685249@192.168.10.1:5070> > Content-Length: 0 > > > I don't have a lot of experience with mediaproxy, and I have some doubts > that such you see they can help me to clarify, inside the file > mediaproxy.ini some options appear which I have configured them but I am not > sure if it is the best way. > > my scenario is the following one: > > <-> UAC<-> NAT <-> ADSL <-> Internet <-> > eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan ( > 192.168.11.1) <-> UAC > > > [MediaProxy] > > start = yes > socket = /var/run/mediaproxy.sock > group = openser > listen = None > allow = None > proxyIP = x.x.x.x (public ip) > ;portRange = 60000:65000 > portRange = 35000:65000 > TOS = 0xb8 > idleTimeout = 60 > holdTimeout = 3600 > forceClose = 0 > > [Accounting] > ; one of none, radius or database > accounting = none > > [Database] > user = dbuser > password = dbpass > host = dbhost > database = radius > table = radacct > > [Radius] > secret = secret > server = localhost > authport = 1812 > acctport = 1813 > dictionaries = /etc/radiusclient-ng/dictionary, > /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary > retries = 2 > timeout = 3 > > > > > this couple of you line inside the openser, I don't still understand them > according to the guide of ser getting started they are for asymmetric > clients, but I don't find an example > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > somebody that can give me a good help... > > regards > > rickygm > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

1 0

[Kamailio-Users] Fwd: Re: How to enable local_route feature ?
by Henning Westerholt 29 Oct '08

29 Oct '08

On Wednesday 29 October 2008, Pascal Maugeri wrote: (you can reach the kamailio list at lists.kamailio.org, the lists at openser.org goes to the opensips lists) > Following your instructions I have enable the compilation defined in > Makefile.defs and recompile everything after a make clean. One capture of > the compilation for you to see the define is enable: > > [...] > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott > -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK > [..] > checks.c -o checks.o > [...] > > When I try to start the proxy it complains about: > > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > syntax error > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > invalid local_route statement > > As you announced this feature for OpenSER 1.4 release series, I assume this > is available for kamailio v1.4.2 ? Hi Pascal, i just tried it with the trunk (which should show the same behaviour with regards to this feature). My compilation accepts local_route { [..] } in the configuration. Can you quote the relevant part of your configuration? Cheers, Henning

1 0

[Serusers] Error logs for SER
by Shruti Sanadhya 29 Oct '08

29 Oct '08

Hi, I am trying to setup a SER on Ubuntu 8.04 (Hardy heron). I have been following the starter's guide for configuring my SER but I do not know where to look for the error logs. I just get a 'bad config file' error. Where can I get more details? Thanks

2 1

[Kamailio-Users] error when initilazing "uac.so"
by Alex R.S.M 28 Oct '08

28 Oct '08

Hi, I added "uac.so" module to be able to use "uac_replace_from" but got the following error. I tried to initialized the module params I could think of. I am wondering what I am missing . Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:avpops:avpops_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:uac:mod_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:uac:mod_init: 'append_fromtag' RR param is not enabled! - required by AUTO restore mode Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:init_mod: failed to initialize module uac Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:main: error while initializing modules Thanks, Alex _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 28 Oct '08

28 Oct '08

Thanks I'm now trying Zoiper, and it works perfectly, I have solved lot of problems that I thought that were because of my Proxy configuration, but they were because of my X-Lite. Now I will try Twinkle. Thanks for your comments. David Cespedes ----- Mensaje original ----- De: Victor Pascual Ávila <victor.pascual.avila(a)gmail.com> Fecha: Sábado, Octubre 25, 2008 12:51 pm Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > On Sat, Oct 25, 2008 at 12:09 PM, Iñaki Baz Castillo > <ibc(a)aliax.net> wrote: > > El Sábado, 25 de Octubre de 2008, ingdavidcespedes(a)cable.net.co > escribió:> > >> since we don't have an upgrade of X-lite for linux > >> systems, could you recommend me another Softphone. > > > > Always Twinkle softphone, the best one ;) > > I agree-- Twinkle is available for Linux only. > -- > Victor Pascual Ávila > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] My contrib to the project - Simple UI
by Graham Wooden 28 Oct '08

28 Oct '08

Hi list, After seeing that CDRTool was a little too powerful (and complicated for that matter) I decided to take it's CDR gathering and add a "simple" web UI to it allowing me to maintain my SIP users and various settings (some aren't shown in this version right now). Let me preface this - this is very plain, meaning it's not stylized. No CSS, no fancy layouts, etc. Mainly because I have this within my existing UI. However, I stripped it out and is functional as described. Again, for the CDR stuff it does require to have the RADIUS accounting and MySQL setup from CDRTool. Link: https://www.leasedminds.com/ui/ If people see value with this, I will be more than happy to work with someone to make it "stylized" and more featured pack. Enjoy and thanks, -graham

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[Kamailio-Users] Proxying media on reinvite
by CSB 28 Oct '08

28 Oct '08

We have the following scenario: Nated UAC receives INVITE Call is established, we proxy audio Far end sends Re-INVITE (to inform UAC to use UDPTL) We don't proxy audio so call fails So I need to understand how the appropriate way to force us to proxy the audio when the far end of a reinvite is NATed. Relevant parts of openser.cfg (OpenSER 1.3.2): We're using mediaproxy. modparam("usrloc", "nat_bflag", 6) route { # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("64.38.1.1:5060;nat=yes"); # insert IP address #xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; #if (method!="REGISTER") { # record_route(); #}; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } The re-invite looks like this: INVITE sip:6422334455@58.28.1.1:5061 SIP/2.0 Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 Contact: <sip:022998877@202.180.1.1> Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE User-Agent: UAS Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 349 v=0 o=root 5466 5467 IN IP4 202.180.1.1 s=session c=IN IP4 202.180.1.1 t=0 0 m=image 4724 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy Our OK looks like this: SIP/2.0 200 OK To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport=5060 Record-Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> Contact: 6422334455 <sip:6422334455@58.28.1.1:5061> Server: Linksys/SPA2102-5.1.9 Content-Length: 265 Content-Type: application/sdp v=0 o=- 69685 69685 IN IP4 192.168.1.113 s=- c=IN IP4 192.168.1.113 t=0 0 m=image 16386 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy Unfortunately this code doesn't result in the c record of the OK being updated to the server IP (it is still the private IP of the UAC). In the loose route section, the re-invite doesn't pass (client_nat_test("3") || search("^Route:.*;nat=yes")) so we don't use mediaproxy. In the on-reply route bflag(6) is not set. I don't understand why bflag(6) is not set since the r-uri is to a registered user that has bflag(6) set (at least that's how I understand it). Could anyone point me in the right direction? Thanks Cameron

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[Kamailio-Users] [presence] Modifying NOTIFY headers set
by Pascal Maugeri 27 Oct '08

27 Oct '08

Hi I need to add a SIP header to outgoing NOTIFY messages, just before it is sent to remote users. Is that possible to do that without using the local_route section (using openser 1.3) ? Regards, Pascal

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[Kamailio-Users] [dispatcher] configure the sending of OPTIONS pings
by Pascal Maugeri 27 Oct '08

27 Oct '08

Hi How do you configure dispatcher to send OPTIONS ping to load-balanced proxies ? I'm using kamailio 1.4.2 with this of configuration: [...] modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_ping_from", "sip:fend1@10.132.184.33<sip%3Afend1(a)10.132.184.33> ") modparam("dispatcher", "ds_probing_threshhold", 3) [...] The load-balancing works properly : I see the traffic being sent to my proxies. But I don't see any SIP OPTIONS packets. I've read in module doc "This method is only available, if compiled with the probing of failed gateways enabled. " but I can't find how to activate this on compilation time, so I'm not sure if I have this feature or not enable. Regards, Pascal

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[Kamailio-Users] i want to stop second registration request if user already registered
by Mian M Asif 27 Oct '08

27 Oct '08

Hi All, i want to stop second registration request if use already registered i am using below code but getting no any response. please help me how to solve this problem. if (method=="REGISTER") { if (registered("location") ) { xlog("L_NOTICE","Asif: lookup(location) Passed\n"); sl_send_reply("699", "User Not Found"); return; } route(2); return; route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_NOTICE","Danish: Entering route[2]\n"); # if (registered("location")) { # xlog("L_NOTICE","Asif: lookup(location) failed\n"); # sl_send_reply("404", "User Not Found"); # return; # }; xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], RecvdIPaddr[$Ri]\n"); if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); setflag(6); setbflag(6); # if you want OPTIONS natpings uncomment next setbflag(7); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize("abc.com")) { www_challenge("abc.com", "1"); }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } } regards, Asif

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[Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 25 Oct '08

25 Oct '08

Hi I'm using Xlite 2.0 release 1105d, and when I try to REGISTER to IP:5070, it doesn't let me. In another Windows computer I have Xlite 3.0 and there you can user an IP:Port Address to register to. Does anyone know If it is a problem from version 2.0 or I am doing something wrong. If it is a problem of 2.0 version, and since we don't have an upgrade of X-lite for linux systems, could you recommend me another Softphone. Thanks David Cespedes

3 2

[Kamailio-Users] Stateless mode for sip load balancer
by Kaeman Chris 25 Oct '08

25 Oct '08

Hello, I have opensip working as a sip load balancer which balances load between 8 asterisk servers running 4 instances on each. The opensip task is just to load balance the incoming requests for 32 asterisk instances on 8 boxes. Can i run it in the stateless mode? Anything that i should be aware about which will prohibit me to run it on the stateless mode. Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 2

Re: [Kamailio-Users] kamailio with tls
by ingdavidcespedes＠cable.net.co 24 Oct '08

24 Oct '08

I knowed that you can have TLS connection between UA and other Proxy Servers, but I didn't know that Kamailio supports it, thanks for your comment. So maybe Paulos's problem is that his Client is not trying to REGISTER to the correct port (5061) of his Kamailio. ----- Mensaje original ----- De: Klaus Darilion <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 12:45 pm Asunto: Re: [Kamailio-Users] kamailio with tls > (cc'ed to the mailinglist) > > ingdavidcespedes(a)cable.net.co schrieb: > > Enable TLS in Kamailio is only for connection between other Kamailio > > (SIP Proxy Servers), it doesn't have anything to do with UA, or > am I > > wrong? . As I understand, you can not implement TLS between UA's and > > Kamailio. > > Of course you can. If the SIP client supports TLS you can also use > TLS > between UA and Kamailio - no problem. > > I use it with eyebeam clients and SNOM phones. Also pjsip supports > TLS, > for example you can test it using QjSimple. > http://www.ipcom.at/index.php?id=560 > > regards > klaus > > > > > > ----- Mensaje original ----- De: Klaus Darilion > > <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 > 11:04> am Asunto: Re: [Kamailio-Users] kamailio with tls > > > >> > >> paulo leonardo schrieb: > >>> Hi, > >>> > >>> I would like a litle help :D! > >>> > >>> I installed kamailio and everything is ok :D! But i want to use > >> TLS, but > >>> when i setup TLS in kamailio, don't work the REGISTER (i can't > >> register > >>> my sofphone ...). And i compliled kamilio with TLS!!! when i > >> comment the > >>> cofigurantion TLS and use port 5060 works! > >> > >> So, "what" does not work? > >> > >> - Does Kamailio start (ps -Alf|grep kamailio)? - Does K listen to > >> the specified TLS socket (netstat -anp|grep kama)? - which client > >> do you use? - is a TCP connection set up? - is a TLS handshake > >> happening (ssldump)? - what is in the logfile of Kamailio? ..... > >> > >> Your error description is to short! > >> > >> regards klaus > >> > >> > >> > >> > >>> below is my configuration > >>> > >>> ----------------------------------------- disable_tls = no listen > >>> = tls:192.168.170.101:5061 <http://192.168.170.101:5061> > >>> tls_verify_server = 1 tls_verify_client = 1 > >>> tls_require_client_certificate = 1 tls_method = TLSv1 > >>> tls_certificate = > >>> "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key > >>> = "/usr/local/etc/kamailio/tls/user/user- > >> privkey.pem"> tls_ca_list > >>> --------------------------------------- > >>> > >>> if (!www_authorize("192.168.170.101 <http://192.168.170.101>", > >>> "subscriber")) { www_challenge("192.168.170.101 > >> <http://192.168.170.101>", "0"); > >>> exit; } > >>> > >>> ---------------------------------------- > >>> > >>> root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf > >>> README request.conf rootCA user user.conf > >>> > >>> tls/rootCA: cacert.pem certs index.txt private serial > >>> > >>> tls/rootCA/certs: > >>> > >>> tls/rootCA/private: cakey.pem > >>> > >>> tls/user: user-calist.pem user-cert.pem user-cert_req.pem > >>> user-privkey.pem > >>> > >>> > >>> thanks!!! > >>> > >>> > >>> > >>> ---------------------------------------------------------------- > -- > >>> > >> ------ > >>> _______________________________________________ Users mailing > >>> list Users(a)lists.kamailio.org > >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> _______________________________________________ Users mailing > list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Problems with CDRTool installation!
by Nuno Marques 24 Oct '08

24 Oct '08

Hi, I'm trying to follow the INSTALL.txt of the CDRTool to accomplish a nice installation of OpenSER 1.3.2 + Mediaproxy + SerMyAdmin + Asterisk + CDRTool + FreeRadius. All went well until i started CDRTool installation... it's a little confusing for a newbie like me but... all the steps of the installation seem to went well. When i go to the browser to look for the CDRTool portal it gives me this error: Database error: pconnect as cdradmin to sipdb failed. Session halted. I thought it was a mysql users problem... so i went to the shell and tried to enter mysql with login and password for cdradmin, and... No problem. It worked fine. Mysql accepted the user login/password. Is there someone who can tell me where can i look for errors... I'm really stuck on this part... Thanks in advance, Nuno

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[Kamailio-Users] draft sip-outbound
by paulo leonardo 24 Oct '08

24 Oct '08

Hi, I would like to know if kamailio implements draft sip-outbound? Can i use draft sip-outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 thanks!

2 1

Re: [Kamailio-Users] kamailio with tls
by Klaus Darilion 24 Oct '08

24 Oct '08

(cc'ed to the mailinglist) ingdavidcespedes(a)cable.net.co schrieb: > Enable TLS in Kamailio is only for connection between other Kamailio > (SIP Proxy Servers), it doesn't have anything to do with UA, or am I > wrong? . As I understand, you can not implement TLS between UA's and > Kamailio. Of course you can. If the SIP client supports TLS you can also use TLS between UA and Kamailio - no problem. I use it with eyebeam clients and SNOM phones. Also pjsip supports TLS, for example you can test it using QjSimple. http://www.ipcom.at/index.php?id=560 regards klaus > > ----- Mensaje original ----- De: Klaus Darilion > <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 11:04 > am Asunto: Re: [Kamailio-Users] kamailio with tls > >> >> paulo leonardo schrieb: >>> Hi, >>> >>> I would like a litle help :D! >>> >>> I installed kamailio and everything is ok :D! But i want to use >> TLS, but >>> when i setup TLS in kamailio, don't work the REGISTER (i can't >> register >>> my sofphone ...). And i compliled kamilio with TLS!!! when i >> comment the >>> cofigurantion TLS and use port 5060 works! >> >> So, "what" does not work? >> >> - Does Kamailio start (ps -Alf|grep kamailio)? - Does K listen to >> the specified TLS socket (netstat -anp|grep kama)? - which client >> do you use? - is a TCP connection set up? - is a TLS handshake >> happening (ssldump)? - what is in the logfile of Kamailio? ..... >> >> Your error description is to short! >> >> regards klaus >> >> >> >> >>> below is my configuration >>> >>> ----------------------------------------- disable_tls = no listen >>> = tls:192.168.170.101:5061 <http://192.168.170.101:5061> >>> tls_verify_server = 1 tls_verify_client = 1 >>> tls_require_client_certificate = 1 tls_method = TLSv1 >>> tls_certificate = >>> "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key >>> = "/usr/local/etc/kamailio/tls/user/user- >> privkey.pem"> tls_ca_list >>> --------------------------------------- >>> >>> if (!www_authorize("192.168.170.101 <http://192.168.170.101>", >>> "subscriber")) { www_challenge("192.168.170.101 >> <http://192.168.170.101>", "0"); >>> exit; } >>> >>> ---------------------------------------- >>> >>> root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf >>> README request.conf rootCA user user.conf >>> >>> tls/rootCA: cacert.pem certs index.txt private serial >>> >>> tls/rootCA/certs: >>> >>> tls/rootCA/private: cakey.pem >>> >>> tls/user: user-calist.pem user-cert.pem user-cert_req.pem >>> user-privkey.pem >>> >>> >>> thanks!!! >>> >>> >>> >>> ------------------------------------------------------------------ >>> >> ------ >>> _______________________________________________ Users mailing >>> list Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

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[Kamailio-Users] kamailio with tls
by paulo leonardo 24 Oct '08

24 Oct '08

Hi, I would like a litle help :D! I installed kamailio and everything is ok :D! But i want to use TLS, but when i setup TLS in kamailio, don't work the REGISTER (i can't register my sofphone ...). And i compliled kamilio with TLS!!! when i comment the cofigurantion TLS and use port 5060 works! below is my configuration ----------------------------------------- disable_tls = no listen = tls:192.168.170.101:5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list --------------------------------------- if (!www_authorize("192.168.170.101", "subscriber")) { www_challenge("192.168.170.101", "0"); exit; } ---------------------------------------- root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf README request.conf rootCA user user.conf tls/rootCA: cacert.pem certs index.txt private serial tls/rootCA/certs: tls/rootCA/private: cakey.pem tls/user: user-calist.pem user-cert.pem user-cert_req.pem user-privkey.pem thanks!!!

2 1

[Serusers] Asterisk -> SER one way audio call issue
by Thorsten 24 Oct '08

24 Oct '08

Hi guys does someone have experience about the interaction of asterisk and SER. I have a strange behavior. I terminate a call from an Asterisk to SER and then to a interconnection partner which runs also on asterisk. Whenever I make a call after a long (3-5 mins) idle time I got a one way audio call because the interconnection partner sends it back to SER instead of asterisk. But the call after works like a charm. And this happens all the time. This is my SER config # ----------- global configuration parameters ------------------------ #debug=4 debug=2 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=82.98.89.140:5060 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/ser/current/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/current/lib/ser/modules/exec.so" loadmodule "/usr/local/ser/current/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/current/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/current/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/current/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/current/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/current/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/current/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/group.so" loadmodule "/usr/local/ser/current/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/current/lib/ser/modules/auth.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/current/lib/ser/modules/uri.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="ACK") { route(1); break; } if (method=="REGISTER") { save("location"); break; }; # only forward if from our nat ip if (method=="INVITE" && (src_ip==82.98.89.146 || src_ip==82.98.89.147 || src_ip==82.98.89.129)) { #record_route(); if (uri =~ "sip:[0-9]*@*") { #record_route(); route(3); break; } } else { sl_send_reply("401", "Unauthorized"); }; if (method=="CANCEL") { route(1); break; }; if (method=="BYE") { route(1); break; } lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3] { # strip the first 4 characters 1991 #strip(4); #rewritehost("62.134.16.104"); rewritehostport("62.134.16.104:5060"); #t_relay_to_udp("62.134.16.104", "5060"); xlog("L_WARN", "[%Tf]: Call forwarded to BT from: [%fu] to: [%tu] next: [%ds]\r\n"); route(1); break; } The asterisk dialplan looks like this ; route everything to SER exten => _X.,1,Noop(${SIP_HEADER(FROM)}) exten => _X.,n,Set(SIP_CODEC="gsm") exten => _X.,n,Dial(sip/${EXTEN}(a)82.98.89.140) exten => _X.,n,Hangup() The asterisk server has the public IP 82.98.89.146 and SER 82.98.89.140. BT which is our interconnection partner has the IP 62.134.16.104. Could someone help me with this? Thanks Ciao Thorsten Kohlhepp

1 0

[Serusers] registrar: interesting, however rare, bug..
by Alexandre Snarskii 24 Oct '08

24 Oct '08

Hi! This week we saw interesing problem with one of our clients: initially he was not able to configure his CCME to register on our proxy, so we just added static location serctl ul add ..., that's our classic workaround for CCME and another not-so-smart admins. But when he finally configured registration - things gone nuts: clients asks for registration: Date: Fri, 24 Oct 2008 13:16:29 GMT Timestamp: 1224854189 CSeq: 2905600 REGISTER Contact: <sip:NNNNNNN@AA.BBB.CCC.DD:5060> Expires: 3600 Authorization: Digest .... and ser replies with 200 OK, but with Contact: field filled with his static registration: Contact: <sip:NNNNNNN@AA.BBB.CCC.DD>;q=1;expires=4293498731;received="" As you can see, expires value is 4293498731, which means that it will be valid for about 136 years, just sixteen days short than 2^32 seconds... So, by RFC, client's router should wait for ~100 years before attempting registration refresh :) But CCME, i suppose, thinks different, it just adds received expires value to current time, got uint32 overflow and result is now()-16 days. As a result CCME thinks that registration is always expired and tries to re-register immediately... And does so hundredths times per second, creating unnecessary extra load to our proxy and database backend.... Why that is so: modules/registrar/reply.c calculates expires= as int2str((int)(c->expires - act_time), &len) but, c->expires can be less than act_time, so, result will be negative integer (translated to unsigned int on call to int2str, and this unsigned int will have higher bit set on, so it will be big one). Well, there are check if(VALID_CONTACT(c, act_time)) which filters out most of the cases where с->expires <= act_time, but for 'static' registrations these checks are off: ((c->expires > act_time) || (c->flags & FL_PERMANENT)) is always true in case of 'static' contact because of FL_PERMANENT. For now I fixed it with following patch (not perfect, i suppose, it just uses default_expires value for negative expires time, but it was easy to write and it works for me): --- reply.c.orig Thu Jan 13 11:48:08 2005 +++ reply.c Fri Oct 24 14:08:13 2008 @@ -163,7 +163,11 @@ memcpy(p, EXPIRES_PARAM, EXPIRES_PARAM_LEN); p += EXPIRES_PARAM_LEN; - cp = int2str((int)(c->expires - act_time), &len); + if (c->expires > act_time) { + cp = int2str((int)(c->expires - act_time), &len); + } else { + cp = int2str(default_expires, &len); + } memcpy(p, cp, len); p += len; Environment: ser-0.9.6/i386/FreeBSD (ser-2.0 has the same code, so it may be affected too). The only tuning of usrloc and registrar modules is db_url and dbmode = 2.

1 0

Re: [Kamailio-Users] Users Digest, Vol 41, Issue 68
by Mian M Asif 24 Oct '08

24 Oct '08

Dear Abalashov, What do you mean by "no[t] getting any response? is mean this function is not working please let me know i am doing right or wrong, dear Arif, exit also is not working cheers :). regards, Asif > Today's Topics: > > 1. Re: Architecture with load-balancing (dispatcher module) > (Ovidiu Sas) > 2. i want to stop second registration request if user already > registered (Mian M Asif) > 3. Re: i want to stop second registration request if user > already registered (Alex Balashov) > 4. Re: i want to stop second registration request if user > already registered (arif.zaman) > 5. Re: Anyone doing CNAM lookups? (Klaus Darilion) > 6. Kamailio v1.4.2 Released (Daniel-Constantin Mierla) > > > Message: 2 > Date: Thu, 23 Oct 2008 19:49:21 +0500 > From: "Mian M Asif" <asif44pk(a)gmail.com> > Subject: [Kamailio-Users] i want to stop second registration request > if user already registered > To: users(a)lists.kamailio.org > Message-ID: > <b005c33b0810230749v1b00bf9auabb97cfc17e517c6(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi All, > i want to stop second registration request if use already registered i > am using below code but getting no any response. please help me how to > solve this problem. > > if (method=="REGISTER") { > if (registered("location") ) { > xlog("L_NOTICE","Asif: lookup(location) Passed\n"); > sl_send_reply("699", "User Not Found"); > return; > } > route(2); > return; > > route[2] { > > # ----------------------------------------------------------------- > # REGISTER Message Handler > # ---------------------------------------------------------------- > xlog("L_NOTICE","Danish: Entering route[2]\n"); > # if (registered("location")) { > # xlog("L_NOTICE","Asif: lookup(location) failed\n"); > # sl_send_reply("404", "User Not Found"); > # return; > # }; > > xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], > RecvdIPaddr[$Ri]\n"); > if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { > xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); > setflag(6); > setbflag(6); > # if you want OPTIONS natpings uncomment next > setbflag(7); > fix_nated_register(); > force_rport(); > }; > > sl_send_reply("100", "Trying"); > > if (!radius_www_authorize("abc.com")) { > www_challenge("abc.com", "1"); > }; > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > > } > } > > regards, > Asif > > > > ------------------------------ > > Message: 3 > Date: Thu, 23 Oct 2008 10:51:30 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [Kamailio-Users] i want to stop second registration > request if user already registered > To: Mian M Asif <asif44pk(a)gmail.com> > Cc: users(a)lists.kamailio.org > Message-ID: <49008F72.1060700(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > What do you mean by "no[t] getting any response?" > > Mian M Asif wrote: > >> Hi All, >> i want to stop second registration request if use already registered i >> am using below code but getting no any response. please help me how to >> solve this problem. >> >> if (method=="REGISTER") { >> if (registered("location") ) { >> xlog("L_NOTICE","Asif: lookup(location) Passed\n"); >> sl_send_reply("699", "User Not Found"); >> return; >> } >> route(2); >> return; >> >> route[2] { >> >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ---------------------------------------------------------------- >> xlog("L_NOTICE","Danish: Entering route[2]\n"); >> # if (registered("location")) { >> # xlog("L_NOTICE","Asif: lookup(location) failed\n"); >> # sl_send_reply("404", "User Not Found"); >> # return; >> # }; >> >> xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], >> RecvdIPaddr[$Ri]\n"); >> if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { >> xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); >> setflag(6); >> setbflag(6); >> # if you want OPTIONS natpings uncomment next >> setbflag(7); >> fix_nated_register(); >> force_rport(); >> }; >> >> sl_send_reply("100", "Trying"); >> >> if (!radius_www_authorize("abc.com")) { >> www_challenge("abc.com", "1"); >> }; >> consume_credentials(); >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> >> } >> } >> >> regards, >> Asif >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 4 > Date: Thu, 23 Oct 2008 07:59:45 -0700 (PDT) > From: "arif.zaman" <arif.zaman(a)brotecs.com> > Subject: Re: [Kamailio-Users] i want to stop second registration > request if user already registered > To: users(a)lists.kamailio.org > Message-ID: <20132764.post(a)talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > > Mian M Asif wrote: >> >> if (method=="REGISTER") { >> if (registered("location") ) { >> xlog("L_NOTICE","Asif: lookup(location) Passed\n"); >> sl_send_reply("699", "User Not Found"); >> return; >> } >> > > Use "exit;" instead of "return;". > > Cheers, > ARIF >

1 0

[Kamailio-Users] Anyone doing CNAM lookups?
by Brian McCrary 24 Oct '08

24 Oct '08

I've been using OpenSER 1.0.1 for the past couple of years, but I now need to add CNAM lookups. It looks like the presence module might be what I need, although from the looks if it, it's made to respond to lookups, instead of actually generate a query to another server for a lookup. I'm looking at doing CNAM lookups with Verisign which look to interface with a SUBSCRIBE message, in which in return Verising sends back a NOTIFY message with one of the fields being "calling-name" which has the CNAM info in it. Has anyone been succesful in implementing such a lookup with Verisign or another CNAM database provider? Brian

3 8

[Serusers] SER without any fork(), It is possible?
by Samuel Ayet San Andres 24 Oct '08

24 Oct '08

Hi all, I´m trying to use SER (SIP Express Router) in a linux enbedded system with uClinux. In my enbedded processor i dont have any MMU so it cant fork(), it uses vfork() that is a cousin of fork but diferent because qhen father vfork a child, stops until the child make execve() or exit. When I run SER on my embedded plataform in a debug mode it shows me: ERROR: main_loop can´t fork(); Then I try to chage all forks for vfork() but it still not working due to the childs never make execve(). And after the explain of the context, here is the question: can I run SER without any child? Can I run SER without making any fork? I mean that only ONE process (witout deamonize) will run alone. I don´t need to manage more than 1 call...I only want to have one process listening to ONE IP. I have try to change the configuration of the ser 'ser.cfg' asi show: fork=no childrens=0 But still without working and i´m not sure if this is the right way to run SER without forking... Thank you for you support. Samuel. --

1 0

[Kamailio-Users] presence network Agent
by suresh tattikota 24 Oct '08

24 Oct '08

Hi all I am working on Presence network Agent. please let me know any one is working on the same . thank you all suresh

2 1

Re: [Kamailio-Users] Record Route and proxy behind static NAT
by Zappasodi Daniele 23 Oct '08

23 Oct '08

Thanks for your reply. I'm agree with your observations. I had already tried to solve the situations as described in the draft (avoiding spiral) and I have built a configuration that seem to work, at least for the basic call, but my solution needs to much tricks and I'm not sure that what I have done is safe and if it works in more complex call flows. I set the IP address in Record Route depending on position public\private of the callee and I substitute the IP address in Record Route for each reply, when it needs. First of all I suppose to know if the caller and callee is private or public (I check if the Request URI domain with RFC 1918) and I don't know if this is right. Then there is the problem to write the record route. I have moved the record route section in a branch route, because here I know the final destination for each branch. Here there is another doubt: I rewrote the record_route_preset function in order to remove the limit of one record route because in parallel forking I set more than record for each transaction, one record route for each branch. What are the consequences? I risk "only" to loose a call or to kill the proxy? I would prefer to leave the record route section at the beginning, but is it possible to substitute the record route after it is added? I remember that is possible to change only the original message and that there are some limitations on the possibility to change an header added during the message processing (I had some segmentation faults when I tryed with a custom header). Just now I'm looking on what to do with a relay from the failure route to a second destination. Before I proceding on this way I would like to know if I'm going on the wrong direction and if someone has found a simpler solution with this kind of scenarios. Regards, Daniele > Date: Wed, 22 Oct 2008 19:02:35 +0200 > From: " I?aki Baz Castillo " <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] Record Route and proxy behind static NAT > Cc: users(a)lists.kamailio.org > Message-ID: > <cc1f582e0810221002i7c95570epf585456a248811d1(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > The solution could be adding two Record-Routes, one with private IP > and the other with public IP, but I'm not sure of how to achieve it > with Kamailio. AFAIK Kamailio *already* does it when the transaction > comes via eth0 and leaves the proxy via eth1, but this is not your > case. > I recommend you reading this draft, and later trying to get it in Kamailio: > http://tools.ietf.org/html/draft-ietf-sip-record-route-fix > (5. Double Record-Routing) ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] Kamailio v1.4.2 Released
by Daniel-Constantin Mierla 23 Oct '08

23 Oct '08

Hello everybody, a new release in 1.4 series is out. Kamailio 1.4.2 is based on the latest version of branch 1.4, including many fixes in code and documentation, therefore those running 1.4.0 or 1.4.1 are advised to upgrade. This is a minor release that includes updates since v1.4.1 - configuration file and database compatibility is preserved. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.2/src/ <http://www.kamailio.org/pub/kamailio/1.4.2/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.2/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.4.2/ Project site at SourceForge.net: http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ <http://www.kamailio.org/docs/modules/1.4.x> What is new in 1.4.x release series is summarized in the announcement of v1.4.0: http://www.kamailio.org/mos/view/KAMAILIO-v1.4.0-Released Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Architecture with load-balancing (dispatcher module)
by Pascal Maugeri 23 Oct '08

23 Oct '08

Hi I was wondering how to achieve an architecture with two or more active load-balancer nodes (kamailio+dispatcher module). I have read how to setup two dispatcher nodes, one node as a master and the other one as a backup. The backup node doesn't process any traffic until master fails. But how to make the traffic being processed by both load-balancers ? In the case for instance the capacity of single node is not enough to process all incoming traffic. Is there any recommended configuration (eg. using frontal application-level swith) Regards, Pascal

4 4

Re: [Kamailio-Users] Architecture with load-balancing (dispatcher module)
by Daniel-Constantin Mierla 23 Oct '08

23 Oct '08

On 10/23/08 15:05, Pascal Maugeri wrote: > > On Thu, Oct 23, 2008 at 1:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > a solution I am using is to do shared IP (vrrp/ucarp). The typical > scenario is active-backup, when one share IP is used. > > But you can do active-active by using two shared IP addresses, so > will be > > server1: active_ip_1 - backup_ip_2 > server2: backup_ip_1 - active_ip_2 > > So each server is active and backs-up the other. This can be > scaled, with more than 2 servers, you need an IP per server, and > you will have a chain of active servers backing up next one. Based > on who is backing up who, you may have one or more servers down > without affecting the service > > You have to take care of balancing the traffic among the IP addresses. > > > OK but how do you share traffic among the IP addresses ? You need a > kind of super load-balancer that load-balance traffic among > load-balancers ? one way you can do it is via DNS. Daniel > -pascal > > > > > Cheers, > Daniel > > > > On 10/23/08 13:16, Alex Balashov wrote: > > Any "frontal application-level switch" would simply have the > same liability. > > Kamailio is a very high-throughput proxy that can handle huge > amounts of call setups per second. I think you can count on that. > > Failing over around the load balancer node to a secondary load > balancer or distributing the traffic among multiple load > balancers is a job best left to the sending endpoint. For > example, a DID origination provider's switch or SBC can be set > up to fail over calls to a different IP endpoint for your SIP > trunk if no response is received within a certain amount of > time. That is how this is typically done. At some point > you've got to say that you've done all you can do, and it's up > to the other side. > > Pascal Maugeri wrote: > > > > Hi > > I was wondering how to achieve an architecture with two or > more active load-balancer nodes (kamailio+dispatcher module). > > I have read how to setup two dispatcher nodes, one node as > a master and the other one as a backup. > > The backup node doesn't process any traffic until master > fails. But how to make the traffic being processed by both > load-balancers ? In the case for instance the capacity of > single node is not enough to process all incoming traffic. > Is there any recommended configuration (eg. using frontal > application-level swith) > > Regards, > Pascal > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Usrloc: cache vs database inconsistencies
by CSB 23 Oct '08

23 Oct '08

I have noticed a consistent difference between the location records OpenSER has cached and those stored in the database. e.g. openserctl ul show --brief database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump brief Domain:: location table=512 records=19 max_slot=2 Compared to: select count(*) from ser.location; +----------+ | count(*) | +----------+ | 10 | +----------+ 1 row in set (0.00 sec) This difference is consistent over time. When I look at the database logs OpenSER is trying to update an existing record in the database which, for some reason, doesn't exist. Our config has the following relevant parameters modparam("usrloc", "db_mode", 2) Openser 1.3.2 Any thoughts on how to prevent this circumstance from occurring and how to synchronise the cache with the db in an orderly manner? I had thought that after 120 seconds (timer_interval default)this would happen but it doesn't seem to. Any advice appreciated. Regards Cameron

2 1

[Kamailio-Users] Hold - core dump
by Marian Piater 23 Oct '08

23 Oct '08

Hello, I tried Kamailio 1.4.1+TLS with Mediaproxy 2.1.0. Everything works fine, but when I place call to Hold (using Hold button), Kamailio sometimes crashed. When I disable mediaproxy, Kamailio works without crash. I tried using engage_media_proxy() without any end_media_proxy or use_media_proxy and I also tried using use_media_proxy and end_media_proxy without engage_media_proxy with that same result. Sometimes Kamailio crash after first place call to hold, sometimes crash after few place call to hold/unhold. Sometimes work fine. OS is Debian Etch. Kamailio generate core dump. Core was generated by `kamailio'. Program terminated with signal 11, Segmentation fault. #0 0xb7d1e19a in strnlen () from /lib/tls/i686/cmov/libc.so.6 (gdb) bt #0 0xb7d1e19a in strnlen () from /lib/tls/i686/cmov/libc.so.6 #1 0xb7cf13f1 in vfprintf () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d0fb01 in vsnprintf () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7cf6945 in snprintf () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7888e10 in end_media_session (callid=<value optimized out>, from_tag={s = 0x2 "", len = 4}, to_tag= {s = 0xbf9ce1b8 "remove\r\ncall_id: \r\nfrom_tag: ?I?E?\177\017openser", len = 0}) at mediaproxy.c:1478 #5 0xb7889147 in __dialog_ended (dlg=0x6, type=2048, _params=0xb7883cbc) at mediaproxy.c:1545 #6 0xb7865378 in run_dlg_callbacks (type=2048, dlg=0xb59e6398, msg=0x0, dir=0, dlg_data=0x0) at dlg_cb.c:252 #7 0xb7876186 in destroy_dlg_table () at dlg_hash.c:118 #8 0xb7861d8e in mod_destroy () at dialog.c:529 #9 0x080b3d41 in destroy_modules () at sr_module.c:381 #10 0x08069b60 in cleanup (show_status=1) at main.c:333 #11 0x0806aaf1 in handle_sigs () at main.c:523 #12 0x0806ecc8 in main (argc=1, argv=0xbf9cecc4) at main.c:865 DEBUG from Kamailio: Oct 22 16:25:34 [7817] DBG:tm:t_relay_to: new transaction fwd'ed Oct 22 16:25:34 [7856] CRITICAL:core:receive_fd: EOF on 15 Oct 22 16:25:34 [7856] DBG:core:handle_ser_child: dead child 8, pid 7817 (shutting down?) Oct 22 16:25:34 [7856] DBG:core:io_watch_del: io_watch_del (0x8175e80, 15, -1, 0x0) fd_no=26 called Oct 22 16:25:34 [7803] INFO:core:handle_sigs: child process 7817 exited by a signal 11 Oct 22 16:25:34 [7803] INFO:core:handle_sigs: core was generated Oct 22 16:25:34 [7803] INFO:core:handle_sigs: terminating due to SIGCHLD Oct 22 16:25:34 [7856] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 144400, used+overhead=169568, free=879008 max used (+overhead)= 169568 dumping free list: hash = 115 fragments no.: 1, unused: 0 bucket size: 920 - 920 (first 920) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 873576) TOTAL: 6 free fragments = 879152 free bytes ----------------------------- Oct 22 16:25:34 [7853] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7850] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7847] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7844] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7829] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127784, free=920792 max used (+overhead)= 127864 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) hash = 62 fragments no.: 1, unused: 0 bucket size: 496 - 496 (first 496) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 8 free fragments = 920936 free bytes ----------------------------- Oct 22 16:25:34 [7826] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127776, free=920800 max used (+overhead)= 127784 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920944 free bytes ----------------------------- Oct 22 16:25:34 [7823] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127776, free=920800 max used (+overhead)= 127784 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920944 free bytes ----------------------------- Oct 22 16:25:34 [7820] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127768, free=920808 max used (+overhead)= 127768 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920952 free bytes ----------------------------- Oct 22 16:25:34 [7814] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=128304, free=920272 max used (+overhead)= 132536 dumping free list: hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 25, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 11, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 17 fragments no.: 1, unused: 0 bucket size: 136 - 136 (first 136) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 40 fragments no.: 1, unused: 0 bucket size: 320 - 320 (first 320) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) hash = 94 fragments no.: 1, unused: 0 bucket size: 752 - 752 (first 752) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 914936) TOTAL: 68 free fragments = 920416 free bytes ----------------------------- Oct 22 16:25:34 [7811] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=128240, free=920336 max used (+overhead)= 132296 dumping free list: hash = 1 fragments no.: 5, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 4, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 19, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 12, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 1, unused: 0 bucket size: 48 - 48 (first 48) hash = 8 fragments no.: 1, unused: 0 bucket size: 64 - 64 (first 64) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 58 fragments no.: 1, unused: 0 bucket size: 464 - 464 (first 464) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 60 free fragments = 920480 free bytes ----------------------------- Oct 22 16:25:34 [7841] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 111616, used+overhead=136744, free=911832 max used (+overhead)= 136744 dumping free list: hash = 118 fragments no.: 1, unused: 0 bucket size: 944 - 944 (first 944) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 907496) TOTAL: 5 free fragments = 911976 free bytes ----------------------------- Oct 22 16:25:34 [7832] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7807] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7806] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7805] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7804] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102744, used+overhead=128088, free=920488 max used (+overhead)= 128088 dumping free list: hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) hash = 61 fragments no.: 1, unused: 0 bucket size: 488 - 488 (first 488) hash = 101 fragments no.: 1, unused: 0 bucket size: 808 - 808 (first 808) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920632 free bytes ----------------------------- Oct 22 16:25:34 [7835] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7838] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7808] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102600, used+overhead=128496, free=920080 max used (+overhead)= 132720 dumping free list: hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 5, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 9, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 31, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 15, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) hash = 8 fragments no.: 2, unused: 0 bucket size: 64 - 64 (first 64) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 12 fragments no.: 1, unused: 0 bucket size: 96 - 96 (first 96) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 46 fragments no.: 1, unused: 0 bucket size: 368 - 368 (first 368) hash = 115 fragments no.: 1, unused: 0 bucket size: 920 - 920 (first 920) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 914768) TOTAL: 82 free fragments = 920224 free bytes ----------------------------- Oct 22 16:25:34 [7803] DBG:dialog:destroy_dlg: destroing dialog 0xb5a063a0 Oct 22 16:25:34 [7803] DBG:dialog:run_dlg_callbacks: dialog=0xb5a063a0, type=2048 Any suggestions? With best regards Marian Piater -- Marian Piater LAM plus s.r.o. odd. hlasové komunikace - VoIP 242 413 000 Do práce 605 267 612 Mobilní telefon marian.piater(a)lam.cz LAM plus s.r.o. Mezi vodami 19 143 00 Praha - Modřany P Prosím, uvědomte si svou zodpovědnost za životní prostředí, než vytisknete tento e-mail.

2 1

[Kamailio-Users] OpenSER ActiveWatcher Table
by George Lee 22 Oct '08

22 Oct '08

Hi Expert, When a SUBSCRIBE message is received, OpenSER creates an entries in the ActiveWatcher table. This is used to generate NOTIFY message to watchers. The problem I have seen is that OpenSER keeps on creating new entries to the ActiveWatchers if the SIP UA deregister, register, deregister, register, etc. for several times. This triggers multiple NOTIFY messages to be sent to watchers. How does OpenSER remove entries in the ActiveWatcher table? It deosn't look like a SUBSCRIBE with expires=0 would delete an entry in the table. Thanks, George

3 3

[Kamailio-Users] URI length
by VoIP Forums www.Go4Calls.com 22 Oct '08

22 Oct '08

Hi All, I was trying to make dialplan if dialed number length >= 8 it should go for PSTN call otherwise local call. I did the following but PSTN is working well if length is greater than 8 but when number comes for local call i am getting the error "Message to Big" if ($(rU{s.len}) >= 8) { rewritehostport("89.21.XXX.XX:5060"); route(1); exit; }else{ lookup("location"); route(1); exit; }; Thank psl suggest us how i can do the local call if it is < to 8 thank you _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

2 1

[Kamailio-Users] Record Route and proxy behind static NAT
by Zappasodi Daniele 22 Oct '08

22 Oct '08

Hello, I have my proxy behind a NAT (static NAT) with SIP clients that are both on my LAN and on public IP. UAC1 Private IP <--> <--> UAC3 PublicIP Proxy <--> NAT <--> WAN UAC2 Private IP <--> <--> UAC4 PublicIP I'm looking for the correct configuration of record route for this scenario. I looked for examples with proxy behind the NAT and I have always found a config section for the record route like this # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("<MY OPENSER PBULIC IP>:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; In this way the proxy uses "record_route_preset" when the caller is private (client_nat_test("3")) and "record_route" when the caller is public, without consider the callee. Consequently the proxy puts in the record route the public IP when the caller is private and the private IP address when the caller is public. I can't understand how it can work: when a client on an public ip calls a client on an public ip, they find in record route the private ip and they will try to send the SIP method (ACK,BYE, reINVITE ecc) to a private IP, so this messages will be lost. This is the beahaviour that I'm observing in my test. Maybe this configuration isn't adapted for my scenario or I miss something else (I have already added advertised_address/port) or I haven't correctly understand the instructions or the Record Route's usage. ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] UUID column in usr_preferences - How unique does it need to be?
by Graham Wooden 22 Oct '08

22 Oct '08

Hi, By looking at the indexes set for the table usr_preferences, it appears that UUID is going to be unique against the attribute column. So if user A has a attribute of custom_parameter_Z and user B has the same attribute, then both UUIDs need to be different, right? UUID can really be anything right? Its set to VARCHAR(64), so heck - a MD5 checksum against say a epoch timestamp would be fine. I just want to confirm how unique each row (in respects to the UUID) it needs to be. Thanks, -graham

2 2

[Kamailio-Users] Registrar performance
by Pascal Maugeri 22 Oct '08

22 Oct '08

Hi I would like to have some rough estimate of the registrar performance (with DB). If there is no figure available, does it make sense to use call performance benchmark like this one http://www.transnexus.com/White%20Papers/OpenSER_RTPproxy_test.htm and extrapolate to registrar performances ? eg. can I say that the SIP REGISTRAR is able to handle >600 registrations per second in the case of TransNexus testbed ? Regards, Pascal

3 3

[Kamailio-Users] Pike module is not working!
by Arif-Uz-Zaman 21 Oct '08

21 Oct '08

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). J Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

1 0

[Kamailio-Users] Pike module is not working!
by Arif-Uz-Zaman 21 Oct '08

21 Oct '08

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). :-) Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

1 0

[Kamailio-Users] Re write any SIP Header Value!
by arif.zaman 21 Oct '08

21 Oct '08

Hi, Is it possible to update any SIP Header (i.e. User-Agent) value before it saving to “location” database? If yes, please help me with sample config. For example: User-Agent value always save as "arif" in location database. Thanks in advance, ARIF -- View this message in context: http://www.nabble.com/Rewrite-any-SIP-Header-Value%21-tp20067253p20067253.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 7

[Kamailio-Users] (moved to kamailio)Why PUA_BLA generates a SUBSCRIBE "Event: dialog" that is never supported by OpenSIPS
by Klaus Darilion 21 Oct '08

21 Oct '08

Iñaki Baz Castillo wrote: > El Domingo, 19 de Octubre de 2008, Iñaki Baz Castillo escribió: >>> Hi, I using PUA_BLA and PUA_DIALOGINFO modules. >>> >>> I can't understand why when the proxy receives an INVITE it generates a >>> SUBSCRIBE to itselfs with: >>> Event: dialog >>> >>> Shouldn't it be: >>> Event: dialog;sla >>> ? > > Well, I've realized that it's because PUA_DIALOG info (Event: dialog). > > And more sorry! since it's just in Kamailio. The pua_dialoginfo modules sends SUBSCRIBE requests to itself? It can not send SUBSCRIBE - only PUBLISH. I guess the SUBSCRIBE is coming from pua_bla regards klaus

2 1

[Kamailio-Users] Remove function
by michel freiha 21 Oct '08

21 Oct '08

Dear All, I have the following function: if($rU =~ "^[+].*") { route(8); exit; } I need to remove the + sign from the dilaed number before sending the call to route(8) Regards

2 1

[Serusers] simple cluster setup with asterisk
by Uwe Kastens 21 Oct '08

21 Oct '08

Hello, Can anybody give me a hint how to start with a very simple cluster setup? I just want to route SIP to 2 different asterisk server. There is no need to care about NAT or call routing. I was wondering if its enough to use the dispatch module. Is it possible with the dispatch module to achieve that the system will reuse the information where a user is registered to route the calls to the same system? BR kiste -- kiste lat: 54.322684, lon: 10.13586

1 0

[Kamailio-Users] t_relay("0x02")
by jh＠tutpro.com 20 Oct '08

20 Oct '08

tm readme on t_relay flag "0x02" says: 0x02 - do not internally send a negative reply in case of forward failure (due internal error, bad RURI, bad message, etc). When a forward failure occurs, no SIP request is relayed and therefore no negative reply or timeout will show up on the failure_route (if one is set). It applies only when the transaction is created. By default one negative reply is sent. Useful if you want to implement a serial forking in case of failure. i don't quite understand the statement: It applies only when the transaction is created. what does "It" refer to? i did some experiments with new lcr implementation and if, for example, tcp connection to a gw cannot be established, then t_relay without "0x02" flag generates 477 reply to caller and, indeed, failure route is not executed. so i need test if t_relay("0x02") failed due to "send failed" and handle selection of next gw in the same route block where t_relay("0x02") was called. this is inconvenient, because there already exists a failure route block that does all that and now its code needs to be at least partially duplicated. or am i missing something and it is somehow possible to get a failure route block executed also on this kind of internally generated negative reply? -- juha

2 2

[Kamailio-Users] $rr and $rr are null
by jh＠tutpro.com 20 Oct '08

20 Oct '08

i have if (t_check_status("403")) { xlog("L_INFO", "DBG: status is 403\n, rr is <$rr> and rs is <$rs>\n"); }; and i get to syslog: Oct 18 19:51:37 localhost /usr/sbin/kamailio[18580]: DBG: status is 403 , rr is <<null>> and rs is <<null>> is this expected behavior? -- juha

2 4

Re: [Kamailio-Users] opensips db
by Alex Balashov 20 Oct '08

20 Oct '08

waleed hassn wrote: > On Mon, Oct 20, 2008 at 5:41 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> waleed hassn wrote: >>> On Mon, Oct 20, 2008 at 4:25 AM, Alex Balashov >>> <abalashov(a)evaristesys.com> wrote: >>>> waleed hassn wrote: >>>>> hi all >>>>> i am new in opensips , how can i set database and add users >>>> Which database, and what users? >>>> >>>> >>>> -- >>>> Alex Balashov >>>> Evariste Systems >>>> Web : http://www.evaristesys.com/ >>>> Tel : (+1) (678) 954-0670 >>>> Direct : (+1) (678) 954-0671 >>>> Mobile : (+1) (706) 338-8599 >>>> >>> sip users and mysql database >>> an error occur when i execute opensipsdb command >> 1) Learn to set up and administer MySQL. >> >> 2) What error? >> >> 3) It might be beneficial to post to the OpenSIPS-Users list. >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> > thanks , can you get me some useful links to begin First result on Google for "mysql tutorial": http://dev.mysql.com/doc/refman/5.0/en/tutorial.html -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

Re: [Kamailio-Users] opensips db
by Alex Balashov 20 Oct '08

20 Oct '08

waleed hassn wrote: > On Mon, Oct 20, 2008 at 4:25 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> waleed hassn wrote: >>> hi all >>> i am new in opensips , how can i set database and add users >> Which database, and what users? >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> > > sip users and mysql database > an error occur when i execute opensipsdb command 1) Learn to set up and administer MySQL. 2) What error? 3) It might be beneficial to post to the OpenSIPS-Users list. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] opensips db
by waleed hassn 20 Oct '08

20 Oct '08

hi all i am new in opensips , how can i set database and add users thanks

2 1

[Kamailio-Users] RLS and presence Server
by mahesh.peddi＠info-spectrum.com 20 Oct '08

20 Oct '08

Hello, I am able to see RLS properly working when RLS and presence functionality is in the same server. after that, I tried to integrate RLS as a separate application server. To forward the message from RLS to presence server, I made below changes in the config file: route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("NOTIFY")) { rls_handle_notify(); t_release(); } if(is_method("SUBSCRIBE")) { $var(ret_code)= rls_handle_subscribe(); if($var(ret_code)== 10) #to forward to presence server t_relay("udp:<ip>:5070"); } if(is_method("PUBLISH")) { #to forward to presence server. t_relay("udp:<ip>:5070"); t_release(); } exit; } because of this configuration, i found two problems: 1. when i am forwarding subscribe message (from RLS to presence server), its not sending the subscribe for all the contacts. Here, i am expecting (sort of) output of rls_handle_subscribe() to be sent to the presence server. Which is not happening. 2. When presence server is sending notify message, it is sending directly to scscf..instead of rls.. Can anybody let me know, how can I get the desired functionality of: 1. UE --> RLS --> presence server (subscribe) 2. Presence server --> RLS --> UE (for notify). If there is any other way to forward message from RLS to presence, please let me know Regards, Mahesh Peddi

1 0

Re: [Kamailio-Users] Check if a user is calling from aregistered device and if not, deny the call!
by Daniel-Constantin Mierla 20 Oct '08

20 Oct '08

Hello, On 10/19/08 14:32, Arif-Uz-Zaman wrote: > Hi, > > Here is a tricky sample config to check if a user is calling from a > registered device and if not, deny the call: > > $var(temp) = $ruri; > $ruri = $fu; > if(!registered("location")) { > sl_send_reply("403","Forbidden auth ID"); > xlog("<ARIF> Not Found\n"); > exit; > } > $ruri = $var(temp); > in the trunk (so available with upcoming 1.5.0) you can do more strict checks. While you are sure an user is registered, you cannot detect if he is calling from registered device or a different one. You just know it has a valid contact in location database. Here is an example to check Contact address in INVITE is same as one in location table: http://openser.blogspot.com/2008/10/registrar-enhancements.html As you get access to all attributes stored for a location record, you can do many other tests, like source IP and port, etc. Cheers, Daniel > Thanks, > ARIF > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Juha Heinanen > Sent: Sunday, October 19, 2008 3:44 PM > To: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] Check if a user is calling from > aregistereddevice and if not, deny the call! > > Arif-Uz-Zaman writes: > > > I've already tested with the following config but no luck. Since the > > following config check only whether callee is registered or not. Not the > > caller. > > > > if(!registered("location")) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > indeed the function should take a pseudo variable that contains aor used > for checking. > > > Is there any other suggestions? > > save r-uri, store caller's aor to r-uri and call the above function. > then restore r-uri. > > -- juha > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] load/next_contacts question
by Klaus Darilion 20 Oct '08

20 Oct '08

Hi Juha and others! I wonder if I understand load/next_contacts(). If I understand it right it can be used for example to make parallel/serial forking after an ENUM lookup (multiple NAPTRs). I think this is the correct usage: if (enum_query()) { if (load_contacts()) { if (next_contacts()) { t_on_failure("1"); t_relay(); } else { sl_reply("500","error adding contacts"); } } else { sl_reply("500","error fetching destination set"); } exit; } But I wonder what if all the branches after enum_query() have the same q value. According to the README: load_contact: If all contacts in the destination set have the same qvalue, load_contacts() does not do anything thus minimizing performance impact of sequential forking capability when it is not needed. Returns 1 if loading of contacts succeeded or there was nothing to do. Thus, next_contacts() will be called although there are no AVPs to fetch. Do I miss something here? regards klaus

2 2

Re: [Kamailio-Users] Check if a user is calling from a registereddevice and if not, deny the call!
by jh＠tutpro.com 20 Oct '08

20 Oct '08

Arif-Uz-Zaman writes: > I've already tested with the following config but no luck. Since the > following config check only whether callee is registered or not. Not the > caller. > > if(!registered("location")) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } indeed the function should take a pseudo variable that contains aor used for checking. > Is there any other suggestions? save r-uri, store caller's aor to r-uri and call the above function. then restore r-uri. -- juha

3 2

[Kamailio-Users] Send MaxCallDuration and UserCredit for freeradius-cdrtool
by Diego Zuaneti Arruda 20 Oct '08

20 Oct '08

Hi all, How I send to freeradius-cdrtool the maxCallDuration and UserCredit? On the /etc/raddb/users I have: 87001(a)mydomain.com.br Auth-Type := Digest, User-Password == "mypassword" Reply-Message = "Authenticated", SIP-AVP = "authentic#1", SIP-AVP = "callMaxDur:60", SIP-AVP = "credit:25" Is this correct ? Thanks. Diego.

3 28

Re: [Kamailio-Users] How to handle DoS attack and OpenSER Pikemodule!
by arif.zaman 20 Oct '08

20 Oct '08

arif.zaman wrote: > > Hello Daniel, > > I've tested according to your suggessions but No Luck. I've configured to > block for 10 minutes and its working but not works for 2 hours. > >> In my case: If the number of SIP messages from a single IP address to >> my SIP Proxy exceeds 200 per minute. Recommended action: Block IP >> for 2 hours. >> > >> modparam("pike", "sampling_time_unit", 60) modparam("pike", >> "reqs_density_per_unit", 200) modparam("pike", "remove_latency", 7200) > Is there any help? Thanks, ARIF -- View this message in context: http://www.nabble.com/How-to-handle-DoS-attack-and-OpenSER-Pike-module%21-t… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] lcr gateway can now be given a hostname
by jh＠tutpro.com 19 Oct '08

19 Oct '08

lcr gateway can now be given a hostname that is used in request-uri when request is sent to the gateway. hostname is not, however, used in forwarding the request. forwarding is still based on gateway's ip address, port, and transport protocol. this is in order to be able to keep on supporting from_gw() and to_gw() functions. -- juha

1 0

[Kamailio-Users] Check if a user is calling from a registered device and if not, deny the call!
by Arif-Uz-Zaman 19 Oct '08

19 Oct '08

Hi, I do like to configure OpenSER in such a way so that it can accept incoming call from a registered user only. After receiving SIP INVITE message, OpenSER will check whether caller is registered or not before forwarding that received INVITE message to make sure only registered user will make call through OpenSER. If Caller is not registered, OpenSER will deny the call. Is it possible to check if a user is calling from a registered device without using any Database? Please help me with sample configuration by considering my interest. Thanks, ARIF

1 0

[Kamailio-Users] Check if a user is calling from a registered device and if not, deny the call!
by jh＠tutpro.com 19 Oct '08

19 Oct '08

Arif-Uz-Zaman writes: > OpenSER will check whether caller is registered or not before forwarding > that received INVITE message to make sure only registered user will make > call through OpenSER. If Caller is not registered, OpenSER will deny the > call. > Is it possible to check if a user is calling from a registered device > without using any Database? Please help me with sample configuration by > considering my interest. arif, i'm quite sure that this same question has come up many times before already. please check the mailing list archives. -- juha

1 0

[Kamailio-Users] OpenSER 1.3.2 + Mediaproxy 1.9.2 + Asterisk 1.4 + CDRTool with FreeRadius
by Nuno Marques 18 Oct '08

18 Oct '08

Hi to all, I need to implement a simple prototype of a Voip system with OpenSER 1.3.2 + Mediaproxy (for NAT traversal) + Asterisk 1.4 (for interconnection with PSTN) + CDRTool (for accounting) on a debian 4 etch operating system. I've already configured OpenSER + Mediaproxy + Asterisk with sucess. I'm missing CDRTool. I have two questions, if some one can help me: 1º which is the CDRTool version compatible with openser 1.3.2 and mediaproxy 1.9.2. Can i use the latest (6.6.10 i supose)? 2º I've read the install file for CDRTool but it is a little confusing (maybe because i'm a newbie to all this). Can anyone show me some literature (or tutorials) to accomplish this installation? Thanks in advance to all, Nuno

1 0

[Kamailio-Users] simple question
by Alex R.S.M 18 Oct '08

18 Oct '08

Hi, I want openSER to act ad a B2B UA, replacing TO, FROM and URI fields. I know "rewritehostport" function allow me to modify URI. What are functions can be used to change TO and FROM fields. Thanks, _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

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[Serusers] About a SIP load balancing document involving SER
by Iñaki Baz Castillo 17 Oct '08

17 Oct '08

Hi, I've found a document about SIP load balancing in iptel.org website and I would like to comment an error I've found in the document. The document is this: "Towards Effective SIP load Balancing" http://www.iptel.org/voipsecurity/doc/14%20-%20Kambourakis%20-%20Towards%20… Page 7 says: -------------------- SIP Proxies insert a VIA header in all the incoming SIP requests So… In case the LB (Load Balancer) is implemented as another proxy all SIP responses will pass through that proxy Request that belong to a specific dialog should not pass through LB -------------------- This is completely true, but next page (8) says: -------------------- Transparency for responses Prevent Load Balancer from inserting a VIA header E.g. in SER utilizing the SEND core command Modify the SIP's Proxy core to ignore the VIA-header added by the Load Balancer ------------------- There is an important error that unfortunatelly I've realized it's very common. Section 18.2.2 of RFC 3261 says clearly that the responses are *always* sent through the same nodes the request came from. So the response should always traverse the load balancer. 1) Load balancer --- (SIP UDP) ---> UAS In this case the UAS would always reply to the *real* source IP (if this is different of the Via "sent-by" then UAS adds "received" parameter and replies there). 2) Load balancer --- (SIP TCP/SCTP) ---> UAS By definition a UAS must reply using the incoming TCP connection. So it's extrange for me that a document about SIP load balancing tries to offer solutions that are not SIP compliant and also unfeasible (UAS will always reply to the real source IP regardless of the Via content). Since the document proposes SER based solutions (using "SEND" command) I'd just like to confirm if I'm completely right, or maybe it's common those not SIP compliant methods by some vendors in order to provide a load balancing solution. Thanks a lot for any comment and best regards. -- Iñaki Baz Castillo

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Re: [Kamailio-Users] In-dialog request correlation without loose routing?
by Alex Balashov 17 Oct '08

17 Oct '08

Klaus Darilion wrote: > So do you perform lookup() also for in_dialog requests? When necessary. Otherwise, A endpoint just provides the URI of the Z-end of the signaling path for the domain and I make special exceptions to relay that as long as I have the Call-ID stored somewhere, which I do. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 4

Re: [Kamailio-Users] In-dialog request correlation without loose routing?
by Alex Balashov 17 Oct '08

17 Oct '08

Klaus Darilion wrote: > Alex Balashov schrieb: >> BTW, I do think it would be a good idea for the dialog module to >> export these functions directly into the script symbols so they can be >> called that way. I do not like to do loose routing unnecessarily / >> when I have no use for it. > > How does your setup work without loose_route? The proxy sees in-dialog > requests only if you record_route. If you do record_route(), you have to > use loose_route for correct routing. You can't have one without the > other (except you do manual routing also for in-dialog requests) That's generally what I always do - manual routing for in-dialog requests. I was not aware that loose_route() is required for correct routing of in-dialog requests when they are record-routed back through the proxy. I think one of the reasons why this may not be much of an issue for me is because my proxy applications generally always have the proxy as the URI domain - I won't relay for !uri == myself. So, after the initial INVITE is rewritten, the UAC/UAS cores on either side send subsequent in-dialog requests to the same URI they sent the INVITE to. I would, of course, be eager to hear any methodological insights you may offer about what I'm doing incorrectly. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] Openser 1.3.2 to Kamailio 1.4.1 Upgrade Problem. Pls HELP.
by George Lee 17 Oct '08

17 Oct '08

Hi, I am upgrading openser 1.3.2 to kamailio 1.4.1 and I am having this problem: Oct 16 18:20:25 dev-ser-01 /sbin/kamailio[6466]: ERROR:core:db_check_table_version: invalid version 2 for table presentity found, expected 3 (check table structure and table "version") Is there any database schema change to the presentity table? If so, how do I upgrade the presentity table? I tried to run the kamdbctl script to upgrade the database and I got access denied with 'root' user. How do I run kamdbctl with 'openser' and 'openserrw' to access mysql database? Also, my database is on a remote host so kamdbctl may not be useful. Thanks, George

2 2

[Kamailio-Users] In-dialog request correlation without loose routing?
by Alex Balashov 17 Oct '08

17 Oct '08

Also, is there a way to get the dialog module to match subsequent in-dialog requests (i.e. re-INVITEs, ACKs, BYEs) without using the dialog correlator field that gets stamped into the Route: header, and therefore requiring the use of loose_route() to read and consume? I thought that switching the matching mode to pure SIP attributes only would do the trick, but it doesn't. They won't match. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND"
by luzango mfupe 17 Oct '08

17 Oct '08

Hi Iak and Victor, The problem is now solved, the Client appeared to be buggy and not my Kamailio config. Cheers, Lu. Message: 4 Date: Thu, 16 Oct 2008 17:26:24 +0200 From: " Victor Pascual ?vila " <victor.pascual.avila(a)gmail.com> Subject: Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND" during INVITE To: " I?aki Baz Castillo " <ibc(a)aliax.net> Cc: users(a)lists.kamailio.org Message-ID: <618e24240810160826l700fbca9gb61e52ed43dc479(a)mail.gmail.com> Content-Type: text/plain; charset=UTF-8 On Wed, Oct 15, 2008 at 10:48 AM, I?aki Baz Castillo <ibc(a)aliax.net> wrote: > 2008/10/15 luzango mfupe <luzango.mfupe(a)gmail.com>: >> >> >> Hi Mates, >> I am testing a Client in Nokia E65 using kamailio 1.3.3. I managed to >> successfull get it Registered but when i attempt to make a call i always get >> the "404 Not Found" message in the INVITE. Strangely enough, the Client >> appears to still be ONLINE in the location table. > > Your conclusion is wrong. A user doen't need to be registered in order > to make a PSTN call (except if you add that logic to your script, that > is not the case). > > So, forget yuor user is registered, it's doesn't matter, and debug why > your dialed number is not matched as a number to the PSTN. I?aki is right here. In addition, "404 Not Found" is not the correct reponse when the destination target is not registered, IMO it should be a 480 instead. -- Victor Pascual ?vila ------------------------------ Message: 5 Date: Thu, 16 Oct 2008 18:23:19 +0200 From: " I?aki Baz Castillo " <ibc(a)aliax.net> Subject: Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND" during INVITE Cc: users(a)lists.kamailio.org Message-ID: <cc1f582e0810160923h5e6f9ffeq2d6bb8452730646b(a)mail.gmail.com> Content-Type: text/plain; charset=UTF-8 2008/10/16 Victor Pascual ?vila <victor.pascual.avila(a)gmail.com>: > In addition, "404 Not Found" is not the correct reponse when the > destination target is not registered, IMO it should be a 480 instead. Yeah!!! 404 should be returned when the proxy/UAS received a request and it's not responsible for it, for example a non existing user (non existing user != non registered user). When a user is not registered, but it does exist in the server, the server must reply "480". -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......." -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

1 0

[Kamailio-Users] Send me an SMS
by Sunkara RaviPrakash 17 Oct '08

17 Oct '08

1 0

[Kamailio-Users] Send me an SMS
by Sunkara RaviPrakash 17 Oct '08

17 Oct '08

1 0

[Serusers] handing of NAT use MediaProxy
by guolingxi＠126.com 17 Oct '08

17 Oct '08

---------- 转发邮件信息 ---------- 发件人："guolingxi <guolingxi(a)126.com>" 发送日期：2008-10-17 13:48:23 收件人：serusers(a)iptel.org 主题： handing of NAT use MediaProxy Hello: I am a chinese student in beijing University.Recently I intrested in SIP,and I found IPTEL is a good place.I enjoy it very much. I have read your "SER-getting started",it is a very wondful guiding.But i am a beginner,i am confused by some question when I want to use SER to hand of NAT Travesal. Could you give me some advice if I want to use SER to set a serve to do a experiment about NAT Travesal.What hardwares must I need?Now I have only some normal computers ,are they Sufficient？Looking forward your advice or Solution，I will very appreciate if you can provide me some concreted document. You can write to me on you convinience,Thank you for your reading!:) a confused student 2008.10.17

1 0

[Kamailio-Users] lcr module enhancements
by jh＠tutpro.com 16 Oct '08

16 Oct '08

i just committed to kamailio trunk an enhanced version of lcr module. changes from kamailio wiki page http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x are as follows: * New high-performance implementation that keeps lcr information in in-memory hash table, whose size can be given in a new module parameter 'lcr_hash_size'. See lcr/README for lcr function execution times. * New 'weight' field in 'gw' table that can be used to assign a gateway a weight among gateways of its group. * Support for prefix_mode=1 has been dropped. * lcr_dump MI function has been split into lcr_gw_dump and lcr_lcr_dump functions. * lcr_reload function is now executed under a lock thus minimizing race conditions. * Regular expressions are now Perl 5.x, instead of POSIX, compatible due to use of PCRE regular expression library. testing and feedback is appreciated. -- juha

1 0

[Kamailio-Users] Client REGISTERED but "404-NOT FOUND" during INVITE
by luzango mfupe 16 Oct '08

16 Oct '08

Hi Mates,I am testing a Client in Nokia E65 using kamailio 1.3.3. I managed to successfull get it Registered but when i attempt to make a call i always get the "404 Not Found" message in the INVITE. Strangely enough, the Client appears to still be ONLINE in the location table. ################################ kamailio config below ############################## if(!is_method("REGISTER")){ if(nat_uac_test("19")){ record_route(";nat=yes"); } else { record_route(); }; }; ##Loose_route packets if (has_totag()) { #sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; #Check authentication of re-invites if(method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setflag(6); }; route(1); } else { sl_send_reply("404","Not here"); } exit; } ##Loose_route packets if (loose_route()) { if(!has_totag()){ sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; }; route(1); }; #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); # # -- 3 -- Determine Request Target # if (method=="REGISTER") { route(2); } else { route(3); }; } route[1] { # # -- 4 -- Forward request to target # # Forward statefully t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { # sl_reply_error(); }; exit; } route[2] { ## Register request handler if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { ## Requests handler if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); # Radius Extra # $avp(s:sip_proxy_ip)="127.0.0.1"; }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if(!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if (client_nat_test("3")) { append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); setflag(7); force_rport(); fix_contact(); }; #unconditional call forward if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } consume_credentials(); #verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound route(10); } else { # -- Inbound to outbound route(11); }; } else { #From an external domain ->do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { #-- Outbound to inbound route(12); } else { # -- Outbound to outbound route(13); }; }; } route[4] { # routing to the public network rewritehostport("xx.xxx.xxx.xx:5065"); #t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } route[6] { # # -- NAT handling -- # if (isflagset(6) || isflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table #Gateway destinations are handled by regular expressions append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:0[1-9][0-9]*@") { if (is_user_in("credentials","local")) { strip(1); prefix("27"); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for local calls"); exit; }; }; if (uri=~"^sip:00[1-9][0-9]*@") { if (is_user_in("credentials","int")) { strip(2); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for international calls"); }; }; if (!lookup("location")) { route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } failure_route[1] { ##-- ##-- If cancelled, exit. ##-- if (t_was_cancelled()) { exit; }; ##-- ##-- If busy send to the e-mail server, prefix the "b" ##-- character to indicate busy. ##-- if (t_check_status("486")) { revert_uri(); prefix("b"); rewritehostport("xx.xxx.xxx.xx:5065"); append_branch(); route(1); exit; }; ##-- ##-- If timeout (408) or unavailable temporarily (480), ##-- prefix the uri with the "u"character to indicate ##-- unanswered and send to the e-mail ##-- sever ##-- if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); rewritehostport("xx.xxx.xxx.xx:5065"); append_branch(); route(1); exit; }; } failure_route[2] { t_on_failure("2"); t_relay(); } onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_contact(); }; if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } ###################################################################### Captured Invite packets below. ##################################################################### . U 2008/12/12 20:41:19.227089 41.23.215.255:5060 -> xx.xxx.xxx.xx:5060 INVITE sip:0123825710@xx.xxx.xxx.xx SIP/2.0. Via: SIP/2.0/UDP 41.23.215.255:5060 ;branch=z9hG4bK5lvrd249rhhc6dbif4j3i48;rport. From: <sip:1974@10.20.11.62 <sip%3A1974(a)10.20.11.62> >;tag=gm8rd21tl9hc68h6f4j2. To: <sip:0123825710@xx.xxx.xxx.xx>. Contact: <sip:1974@41.23.215.255 <sip%3A1974(a)41.23.215.255>>. Accept-Contact: *;+media=audio. CSeq: 953 INVITE. Call-ID: -u6Sy5l7oIf2mTACTYE46_z8sRn-_f. User-Agent: Tuune Mobile. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 172. . v=0. o=- 54865261 0 IN IP4 41.23.215.255. s=Tuune Mobile. c=IN IP4 41.23.215.255. t=0 0. m=audio 5000 RTP/AVP 0 97. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:97 AMR/8000. U 2008/12/12 20:41:19.227945 xx.xxx.xxx.xx:5060 -> 41.23.215.255:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 41.23.215.255:5060 ;branch=z9hG4bK5lvrd249rhhc6dbif4j3i48;rport= 5060. From: <sip:1974@10.20.11.62 <sip%3A1974(a)10.20.11.62> >;tag=gm8rd21tl9hc68h6f4j2. To: <sip:0123825710@xx.xxx.xxx.xx >;tag=329cfeaa6ded039da25ff8cbb8668bd2.bf78. CSeq: 953 INVITE. Call-ID: -u6Sy5l7oIf2mTACTYE46_z8sRn-_f. Server:Tuune Mobile Test. Content-Length: 0. . #################################### It is online ############ 41:/# openserctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=1 max_slot=1 AOR:: 1974 Contact:: sip:1974@41.22.248.8 <sip%3A1974(a)41.22.248.8> Q= Expires:: 932 Callid:: x1ySy9r2oIf1Ogj2Uzeht2FSEcYGxE Cseq:: 957 User-agent:: n/a State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:xx.xxx.xxx.xx:5060 Methods:: 4294967295 FIFO command was: :ul_dump:openser_receiver_15196 I hope some one can give a clue, WBR, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

3 3

[Kamailio-Users] Problems installing serMyAdmin0.9 in OpenSER1.3.2
by Nuno Marques 16 Oct '08

16 Oct '08

Hi, I tried to install the new version of sermyadmin - 0.9 - and it gave me problems. Created a new openser database and notice that the sermyadmin didn't even made any changes to it. Tried with the old sermyadmin 0.7 and everything went well. I didn't made any changes to tomcat... or anything else. As anyone experienced the same problem? Does anyone knows where could i get sermyadmin0.8 to test it? Thanks in advance for any help on this Nuno

2 1

[Kamailio-Users] debian and solaris packages for kamailio and openser available
by Henning Westerholt 16 Oct '08

16 Oct '08

Hello all, thanks to Muhammad Akl, packages and binaries for the lastest stable release of openser on solaris 10 (on i386) are now available on www.kamailio.org in the download area [1, 2]. He also build packages for some older releases, like the previous 1.3 and 1.2 branch based releases. I added debian packages for the 1.3.3 release of openser and 1.4.1 release of kamailio for etch (on i386) as well [3, 4]. Packages of kamailio on solaris will be also appear soon there, they are still a few incompatibilites that needs to be sorted out that prevent the building. We're working on it.. If you want to contribute some packages for your favorite distribution or operating system, just contact me per mail or on the #kamailio IRC channel. Cheers, Henning [1] http://www.kamailio.org/pub/openser/1.3.3/bin/solaris/10/ [2] http://www.kamailio.org/pub/openser/1.3.3/packages/solaris/10/ [3] http://www.kamailio.org/pub/openser/1.3.3/packages/debian/etch/ [4] http://www.kamailio.org/pub/kamailio/1.4.1/packages/debian/etch/ -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] 1.4.x Dialog module vs. :dlg_end_dlg: command
by Martin Vít 16 Oct '08

16 Oct '08

Hello, i'm testing dialog module and trying to account died sessions. I've problem or i've misunderstood the way how to account it. If I send dlg_end_dlg, a dialog session is ended and BYE is sended to both sides. But if the BYE is not confirmed the acc module will never log this. Is there any way to immidiatly log BYE asap dialog ends? thanks for any suggestion MV

2 5

[Kamailio-Users] Change FROM display name
by ALAEDDINE abbech 16 Oct '08

16 Oct '08

Hi, I should change the actual display name in header From with an other name stored in avp(s:display) . I use uac_replace_from("$avp(s:display)","");, but this function add the new name in the end line of From. Exemple after tcpdump: i like change "aladin" with "1111111111" but i have this line in INVITE request, From: aladin <sip:testerser@x.x.x.x:5060>;tag=64204a12d09be0c^M"11111111111" Have you a solution for this pb. Thanks

2 1

[Kamailio-Users] INVITE queue
by sabino frisardi 16 Oct '08

16 Oct '08

Hi all, i'm tryng to enqueue invite messages in a queue and to make the parsing of the invite after the enqueuing. After the parsing i call tmb.t_relay(msg,0,0,0,0,0,0) from a module. The rest of messages of the classic sip call pass trhough the main route. So when i run openser it consume too much memory. Is there any error with the call to t_relay through the binding? Is there any error on the transaction if i call the relay in a module for INVITE message and in main route for NON_INVITE messages? Why the consume of memory increase? Can anyone help me? Thanks for your help. Regards -- Sabino Frisardi

2 1

Re: [Kamailio-Users] Problems installing serMyAdmin0.9 in OpenSER1.3.2
by Nuno Marques 16 Oct '08

16 Oct '08

Hi Muhammad, The only error that appears is when i tried to access http:// <server_ip>:8080/serMyAdmin/ Gives me an error 404 and says that the resource is not available... HTTP Status 404 - ------------------------------ *type* Status report *message* *description* *The requested resource () is not available.* ------------------------------ Apache Tomcat/6.0.18 I've run java -version and my result is: java version "1.5.0_14" Java(TM) 2 Runtime Environment, Standard Edition (build 1.5.0_14-b03) Java HotSpot(TM) Client VM (build 1.5.0_14-b03, mixed mode, sharing) I think my problem is with the version of Java. Going to try to instal java 6. Thanks a lot Muhammad. Nuno 2008/10/16 muhammad akl <muhammad.akl(a)gmail.com> > > Hi Nuno , > > serMyAdmin-0.9 will work with any version of openser and/or kamailio , > > but alot of changes have been added to the NEW BETA version (0.9) and the > changes AFAIK are depending on java 6 could you please append the error that > appeared for you ? > > Regards > > Muhammad >

1 0

[Kamailio-Users] 1.4.x Dialog module vs. :dlg_end_dlg: command
by Martin Vít 16 Oct '08

16 Oct '08

Hello, i'm testing dialog module and trying to account died sessions. I've problem or i've misunderstood the way how to account it. If I send dlg_end_dlg, a dialog session is ended and BYE is sended to both sides. But if the BYE is not confirmed the acc module will never log this. Is there any way to immidiatly log BYE asap dialog ends? thanks for any suggestion MV

1 0

[Kamailio-Users] Dynamic profile creation for dialog module.
by Alex Balashov 16 Oct '08

16 Oct '08

Hello, It'd be really nice to have a way to create profiles used by the dialog module dynamically, ideally in script but MI would work too. The problem is that I am using many different profiles to keep track of the number of open dialogs for a lot of different call formulas, some of whose existence should be user-configurable. But these profiles have to be defined statically as having values using the dialog module's parameter "profiles_with_value," because I do use pseudovariable values with them. Now, of course, one can simply use one or more big profiles and construct pseudovariable values based on what needs to be profiled, e.g. "$rU:bigblend" But whereas the profiles are matched to tracked dialogs using a hash, the values are linearly compared using memcmp() in modules/dialog/dlg_profile.c:is_dlg_in_profile(). So, there is a speed penalty in doing this. Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] How to change shared memory size?
by Pablo Bertuzzi 15 Oct '08

15 Oct '08

Hi Forum, kamailio failed to start I believe due to a large carrierrote table (50K prefixes). Kamailio version is 1.4.1, OS is debian, physical memory installed is 4Gb. How can I change the size? Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: INFO:carrierroute:get_route_tree: found domain 0 Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: INFO:carrierroute:add_route: found route, now adding Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:carrierroute:add_route_rule: out of shared memory Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:carrierroute:prepare_route_tree: could not load routing data Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:carrierroute:mod_init: could not prepare route tree Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:core:init_mod: failed to initialize module carrierroute Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:core:main: error while initializing modules

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[Kamailio-Users] new feature: non-blocking logging
by Henning Westerholt 15 Oct '08

15 Oct '08

Hi all, i just commited support for non-blocking logging to the trunk. This was implemented using the syslog-async functionality from the dnsmasq server [1] from Simon Kelley. The standard Unix syslog() library routine can block waiting for the syslog daemon. On some systems, using a datagram socket for /dev/log avoids endless waits, but on Linux, even this does not work. Try typing killall -SIGSTOP $pid_of_syslog into a root terminal on a non-critical machine. Then generate traffic to get some logs, eventually, the server will stop. This blocking of the syslog daemon can happens because of bugs, or if the daemon waits for external events, like DNS lookups. Instead of blocking, log-lines are buffered in memory. The buffer size is limited and if the buffer overflows log lines are lost. When lines are lost this fact is logged with a message. This functionality can be enabled with setting the define SYSLOG_ASYNC in the Makefile.defs file. The default is the normal log functionality from the system library, nothing changed if the define is not set. I tested this with our internal tests suite, and also the tests in the trunk. You can get the code from the svn repository, we also provide debian packages for several debian releases on devel.kamailio.org. Cheers, Henning [1] http://www.thekelleys.org.uk/dnsmasq/doc.html -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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[Kamailio-Users] CDRTool 6.6.4 with freeradius-cdrtool
by Diego Zuaneti Arruda 15 Oct '08

15 Oct '08

Hi all, Someone has a patch for use freeradius-cdrtool with the latest version of cdrtool ? The stable versions of cdrtool not have multiple commands that were in earlier versions. Thanks. Diego.

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[Kamailio-Users] I want to have both Presence Network Agent and Resource List Service in my presence network.
by suresh tattikota 15 Oct '08

15 Oct '08

Hi, I want to have both Presence Network Agent and Resource List Service in my presence network. As per the architecture, both presence network agent and resource list server can act as the application servers, and they can communicate with presence server. Can anybody let me know, how to make the communication possible between application server and the presence server? Regards, Suresh

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Re: [Kamailio-Users] Interfaceing Presence server with Presence Network Aget(PNA)
by Daniel-Constantin Mierla 15 Oct '08

15 Oct '08

please cc to mailing list. Daniel On 10/15/08 07:51, suresh tattikota wrote: > hi Daniel thank u > > > *PNA*:presence network agent , it collects the presence information of > the network element like *HSS*(Home Subscriber Server) and > *S-CSCF*(serving call session control function) (i.e registered are > unregistered ).Using '*sh'* interface i am collecting these > information through *SNR/SNA,PNR/PNA*(subscribe notify request and > answer,,push notify request/answer ) after getting the PNR i have to > send PUBLISH message to the presence server through *PEN*(its nothing > but publish message )interface , > > My question is how to connect the presence server with presence > network agent . > and how to attach the presence information to the PUBLISH XML body . > > thank you very much > > > On 10/14/08, *Daniel-Constantin Mierla* <miconda(a)gmail.com > <mailto:miconda@gmail.com>> wrote: > > Hello, > > On 10/13/08 12:30, suresh tattikota wrote: > > > hi all > > I am trying to develop the interface between PNA and PS,that > is PEN interface, > for that i am using the one openser 1.2 as the PNA > Application server and Presence server. > > it is recommended that you use kamailio 1.4.1 as presence server, > it has many improvements in this area, openser 1.2 is rather old > and not supported anymore > > > I already developed sh interface between HSS and the PNA. > After getting the PNR request from HSS,PNA has to send the > PUBLISH message to PS > > so here my doubt is > > 1.How to connect the PNA and PS > 2.how to construct PUBLISH XML body > 3.can i use the opensips/openser as presence network agent > > The problem with the questions you ask is that you use a lot of > abbreviations from IMS specs (as they look) . You won't find good > answers unless you get to more details of what you want to > achieve, preferably at SIP level - nobody will spend time to > understand what PS, PEN, PNA are supposed to do. > > Kamailio (OpenSER) can be used as: > - presence server - can handle PUBLISH/SUBSCRIBE messages and send > NOTIFYs > - presence user agent - initiate PUBLISH requests > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com <http://www.asipto.com/> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] New OpenXCAP release 1.0.5
by Adrian Georgescu 15 Oct '08

15 Oct '08

Hello, There are new releases of the open source OpenXCAP server and a Python XCAP library available. They contain many bug fixes and lessons learnt at SIPIT 23, still in progress in Lannion, France. Changelogs are attached: openxcap (1.0.5) unstable; urgency=low * fixed: openxcap with 'openser' backend could be start if enable_publish_xcapdiff=yes * fixed a couple of potential Internal Server Errors * added python-mysqldb to dependencies * added <entry> node to <uniqueness-failure> error as required by RFC * nicer error messages in several cases openxcap (1.0.4) unstable; urgency=low * Disabled authentication for requests in /global/ context, because it didn't work well. It is safe, because the only use for /global/ now is retrieving xcap-caps. openxcap (1.0.3) unstable; urgency=low * Fixed: requesting XCAP root resulted in 500 (now 404) * Fixed: multiple matches resulted in 500 (now 404) * Improved error responses openxcap (1.0.2) unstable; urgency=high * Fixed and improved logging. it now logs much more then before and can be customized * Fixed bug: xcap-caps application was not available under /global/ tree * Added experimental support for sqlite database * Fixed multiple xcap roots/aliases. having more then one root= entry in the config is legal now * Fixed bug that made OpenXCAP dependent on specific MySQLdb versions (those with 'reconnect' parameter) * Fixed pyxml issue: previously, if pyxml was not installed, OpenXCAP would fail with obscure exception on element operation. Now it refuses to start. * Fixed URI parsing * Better error reporting in many cases * Many more bugs fixes openxcap (1.0.1) unstable; urgency=low * Added proper URI parsing, so att_value may contain slashes now * Fixed schemas not to fetch files from the internet on initialization python-xcaplib (1.0.6) unstable; urgency=low * made "index" default filename for xcapclient as well * made --app unnecessary for all types of requests if -i is present python-xcaplib (1.0.5) unstable; urgency=low * changed default filename from index.xml to index * changed default auth to None (support both basic and digest) * removed loading options from environ (XCAP_ROOT etc) * added -a, --account-name option * added --show-config option python-xcaplib (1.0.4) unstable; urgency=low * added --sip-address option (sip_address in the config file) python-xcaplib (1.0.3) unstable; urgency=low * added support for documents in the global context * added --filename parameter to xcapclient python-xcaplib (1.0.2) unstable; urgency=low * removed setuptools from setup.py completely python-xcaplib (1.0.1) unstable; urgency=low * fixed setup.py to use setuptools only if available How to upgrade or install the software For Debian testing or unstable on an i386 architecture there is an official public repository provided by AG Projects. Install the AG Projects debian software signing key: wget http://download.ag-projects.com/agp-debian-gpg.key apt-key add agp-debian-gpg.key Add these lines in etc/apt/sources.list # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main After that, run: apt-get update apt-get install openxcap python-xcaplib For other operating systems download the tar archive from: http://openxcap.org Kind regards, Adrian Georgescu

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Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND"
by luzango mfupe 15 Oct '08

15 Oct '08

> > Hi Castilo, Yea you might be right that my conclusion is wrong but in my case at least a client must be in the ACL in order to be able to make PSTN calls,unregistered clients cant call as well. I tested the same configuration with a X-lite client and it was possible to make calls. > ...debug why your dialed number is not matched as a number to the PSTN. > > Can you explain here a lil bit more?? because the regex part in my config > is working fine it matches all PSTN[Local & Int.] perfectly!! Cheers, Lu. > > > Message: 2 > Date: Wed, 15 Oct 2008 10:48:48 +0200 > From: " I?aki Baz Castillo " <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND" > during INVITE > Cc: users(a)lists.kamailio.org > Message-ID: > <cc1f582e0810150148h1ba9b282t94ae8278c3550e27(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > 2008/10/15 luzango mfupe <luzango.mfupe(a)gmail.com>: > > > > > > Hi Mates, > > I am testing a Client in Nokia E65 using kamailio 1.3.3. I managed to > > successfull get it Registered but when i attempt to make a call i always > get > > the "404 Not Found" message in the INVITE. Strangely enough, the Client > > appears to still be ONLINE in the location table. > > Your conclusion is wrong. A user doen't need to be registered in order > to make a PSTN call (except if you add that logic to your script, that > is not the case). > > So, forget yuor user is registered, it's doesn't matter, and debug why > your dialed number is not matched as a number to the PSTN. > > > -- > I?aki Baz Castillo > <ibc(a)aliax.net> > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 41, Issue 42 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Rouge UAC: location database not updated
by CSB 15 Oct '08

15 Oct '08

I see the following messages in the logs: Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:siptrace:sip_trace: cannot parse FROM header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:maxfwd:is_maxfwd_present: parsing MAX_FORWARD header failed! Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:rr:find_first_route: failed to parse headers Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_uri: cannot parse FROM header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:domain:is_from_local: Error while parsing From header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:build_res_buf_from_sip_req: parse_headers failed Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:sl:sl_send_reply_helper: response building failed Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:pv_get_to_attr: cannot parse To header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:pv_get_from_attr: cannot parse From header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:pv_get_from_attr: cannot parse From header This seems to have been caused by one rogue UAC and has the effect of preventing entries being written to the location database. My question: could anyone suggest how to prevent a UAC which is sending some weird data from causing the server to choke? Thanks Cameron

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Re: [Kamailio-Users] problem starting kamailio
by ingdavidcespedes＠cable.net.co 15 Oct '08

15 Oct '08

1 0

[Kamailio-Users] problem starting kamailio
by Fabian Borot 15 Oct '08

15 Oct '08

hello I installed kamailio, and want to use mysql auth,loaded the mysql module, uncomented the lines needed to use mysql in the .cfg file I ran the openser_mysql.sh script, and when I try to started kamailio and this is what I see in the console: [root@sp1094a sbin]# ./openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': 65.111.185.187 creating database openser ... Install SERWEB tables ?(y/n):n # I tried with and w/o the servweb tables [root@sp1094a sbin]# ./kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed in the var/log/message file this is what I see: Oct 14 21:09:12 sp1094a ./kamailio[29249]: NOTICE:core:main: version: kamailio 1.4.1-notls (i386/linux) Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:main: using 32 Mb shared memory Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:main: using 1 Mb private memory per process Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:sl:mod_init: Initializing StateLess engine Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:tm:mod_init: TM - initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:maxfwd:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:usrloc:ul_init_locks: locks array size 512 Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:registrar:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:textops:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:xlog:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:acc:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:auth:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:auth_db:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 210 kb Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'cflags' in 'field list' Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:core:db_do_query: error while submitting query Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:usrloc:preload_udomain: db_query (1) failed Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:usrloc:child_init: child(1): failed to preload domain 'location' Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:core:init_mod_child: failed to initializing module usrloc, rank 1 Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:core:main_loop: init_child failed for UDP listener Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:handle_sigs: child process 29253 exited normally, status=255 Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:handle_sigs: terminating due to SIGCHLD Oct 14 21:09:12 sp1094a ./kamailio[29264]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29261]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29259]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29257]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29255]: INFO:core:sig_usr: signal 15 received what can that be? txs Fabian <html><DIV> <DIV> </DIV></DIV></html> _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

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Re: [Kamailio-Users] How to handle DoS attack and OpenSER Pike module!
by Daniel-Constantin Mierla 14 Oct '08

14 Oct '08

Hello, On 10/14/08 09:32, Arif-Uz-Zaman wrote: > Hi all, > I need to bother about crazy client by considering “Flood” detection > technique. I can do it by using OpenSER Pike > <http://kamailio.org/docs/modules/1.2.x/pike.html> module which helps > to keep trace of all (or selected ones) incoming request's IP source > and blocks the ones that exceeded some limit. > > In my case: If the number of SIP messages from a single IP address to > my SIP Proxy exceeds *200* per minute. Recommended action: Block IP > for 2 hours. > > I tried with the pike module but I’m little bit confused with > sampling, density, and timeout value. > > Please help me with example configuration by considering my point. > have you tried: modparam("pike", "sampling_time_unit", 60) modparam("pike", "reqs_density_per_unit", 200) modparam("pike", "remove_latency", 7200) Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] Interfaceing Presence server with Presence Network Aget(PNA)
by suresh tattikota 14 Oct '08

14 Oct '08

hi all I am trying to develop the interface between PNA and PS,that is PEN interface, for that i am using the one openser 1.2 as the PNA Application server and Presence server. I already developed sh interface between HSS and the PNA. After getting the PNR request from HSS,PNA has to send the PUBLISH message to PS so here my doubt is 1.How to connect the PNA and PS 2.how to construct PUBLISH XML body 3.can i use the opensips/openser as presence network agent thank you regards suresh

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Re: [Kamailio-Users] what is the best solution NAT problem
by Ricky Gutierrez 14 Oct '08

14 Oct '08

Sunday, October 12, 2008 3:43:16 PM Iñaki Baz Castillo wrote: OpenSer behind a NAT router with dynamic IP? It's really an annoying scenario. Note that when the INVITE/200/(ACK) comes from the LAN 192.168.10.X and goes through OpenSer, you must replace the media IP in the SDP and the IP in "Contact" with the **public** IP of the LAN router, that is a dynamic IP and AFAIK it's not valid to set a domain in the SDP. And when he INVITE/200/(ACK) comes from Internet and goes through OpenSer, you must replace the media IP in the SDP and IP in "Contact" with the eth1 IP of RtpProxy (19.168.10.1). Iñaki I am replacing it for the private ip that I obtain of my router adsl 192.168.1.64, in some examples I read that they replace for the ip it public.. what you try to say is that it replaces the ip that comes from the router for the ip of the lan Also, you need the SIP ports and RtpProxy media ports redirected in the router to the RtpProxy server. that already has it configured in my router ... Anyway, this scenario is not appropiate for a businnes service. you are right, I have a wall of NAT I have open the ports UDP 5060:5065, 10000:20000, 35000:65000 TCP: 5060 Why do you open all these ports? and what do you mean with "open"? don't you mean "redirected"? excuse me, I meant that the ports this redirect to my server PD: A suggestion: Buy an space in a datacenter (a virtual machine could be enough depending on your traffic ammount) and install the OpenSer and RtpProxy decently in a host with public IP. ;) best regards .. rickygm

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[Kamailio-Users] presence module, events
by toly 14 Oct '08

14 Oct '08

Greetings, This is a question to the author: bogdan_iancu, but if anybody can answer, I'd appreciate it. Softphone client is Bria. Here is publish event from ngrep PUBLISH sip:tolyp@webley SIP/2.0. Via: SIP/2.0/TCP 10.50.1.101:45464;branch=z9hG4bK- T 2008/10/13 16:15:38.621949 10.50.1.101:1572 -> 10.50.4.40:5070 [AP] d8754z-697cf1172b7bf515-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:tolyp@10.50.1.101:1572;transport=TCP>. To: "tolyp"<sip:tolyp@webley>. From: "tolyp"<sip:tolyp@webley>;tag=ca65f37d. Call-ID: YmFlNjUxMzEwM2Q4Nzk0MWRhZGUxZmQwMDA5NmE4Nzc.. CSeq: 1 PUBLISH. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/pidf+xml. User-Agent: Bria release 2.4 stamp 49274. Event: presence. Content-Length: 441. . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:tolyp@webley'><tuple id='t5a055764'><status><basic>open</basic></status></tuple><dm:person id='p5e347748'><dm:note>Available</dm:note></dm:person></presence> Here is the code from parser/parse_event.c if ((_e->text.len == PRES_STR_LEN) && !strncasecmp(PRES_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_PRESENCE; } else if ((_e->text.len == PRES_XCAP_CHANGE_STR_LEN) && !strncasecmp(PRES_XCAP_CHANGE_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_XCAP_CHANGE; } else if ((_e->text.len == PRES_WINFO_STR_LEN) && !strncasecmp(PRES_WINFO_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_PRESENCE_WINFO; } else if ((_e->text.len == PRES_SIP_PROFILE_STR_LEN) && !strncasecmp(PRES_SIP_PROFILE_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_SIP_PROFILE; } else if ((_e->text.len == DIALOG_STR_LEN) && !strncasecmp(DIALOG_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_DIALOG; } else if ((_e->text.len == MWI_STR_LEN) && !strncasecmp(MWI_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_MWI; } else { _e->parsed = EVENT_OTHER; } if( (*end)== ';') { str params_str; params_str.s= end+1; end= skip_token(params_str.s, tmp.len- _e->text.len- 1); params_str.len= end- params_str.s; if (parse_params(&params_str, CLASS_ANY, &phooks, &_e->params)<0) return -1; } else _e->params= NULL; It detects presence event correctly, but then if( (*end)== ';') does not find any semicolon and here is the result: 10/13 16:34:57 13030 err ERROR:presence:handle_publish: <publish.c: 508> Missing or unsupported event header field value 10/13 16:34:57 13030 err ERROR:presence:handle_publish: <publish.c: 511> event=[presence] What semicolon it is looking? If I do (*end)== ' ') it still can't parse. Is Bria message has incorrect format? What was the format of message the parser was written for? Thanks a lot, Toly -- View this message in context: http://www.nabble.com/presence-module%2C-events-tp19963494p19963494.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Kamailio-Users] How to handle DoS attack and OpenSER Pike module!
by Arif-Uz-Zaman 14 Oct '08

14 Oct '08

Hi all, I need to bother about crazy client by considering "Flood" detection technique. I can do it by using OpenSER Pike <http://kamailio.org/docs/modules/1.2.x/pike.html> module which helps to keep trace of all (or selected ones) incoming request's IP source and blocks the ones that exceeded some limit. In my case: If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Recommended action: Block IP for 2 hours. I tried with the pike module but I'm little bit confused with sampling, density, and timeout value. Please help me with example configuration by considering my point. Thanks, ARIF

1 0

[Serusers] ser-0.9.6: modules/postgres bugreport & patch
by Alexandre Snarskii 13 Oct '08

13 Oct '08

Hi! I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend is postgresql version 8.1.3. Today I got errors in logfiles, saying: messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]: ERROR:avpops:dbrow2avp: dbrow contains NULL fields The similar problem reported in: http://lists.iptel.org/pipermail/serusers/2005-May/019681.html with much more detailed description of error, database contents and config samples. Patch is trivial, and looks more like a fix to copy'n'paste error: in mysql/val.c function str2val states: if (!_s) { memset(_v, 0, sizeof(db_val_t)); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } VAL_NULL(_v) = 0; and the last line mentions that 'well, that's value is not NULL'. In postgresql/db_val.c, line 182, function str2valp, the same statement is the: if (!_s) { DLOG("str2valp", "got a null value"); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } without explicit notification that this is not-NULL value. More than, nowhere else in this function VAL_NULL(_v) not set to 0. So, if a value _v.val contained anyting but 0 at the function start, resulting value will be threated as NULL despite the fact, that _s is NOT NULL. Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and everyting will be ok. At least for me it's ok for some hours :)

2 3

[Serusers] installing CDRTool on Freebsd
by Mohit Saxena 13 Oct '08

13 Oct '08

Hello Friends, Can any one help me installing the CDR TOOL to integrate with freeradius on freebsd? Any help will be highly appreciated. Thanks Mohit Mohit C. Saxena Sr. Network Administrator Starcomms Plc. www.starcomms.com <blocked::http://www.starcomms.com> Mob: +234-702-800-0709 DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks.

5 7

[Kamailio-Users] fr_anything_at_all_timer.
by Alex Balashov 13 Oct '08

13 Oct '08

It would be really nice to have a feature in TM for even faster failover that is turned off when any reply is received at all, provisional or otherwise. The problem with fr_timer and fr_inv_timer is that they throw on lack of final reply. But sometimes gateways have very lengthy, complicated asynchronous operations that must complete before they can return a final OK. If you are doing failover, the most important thing to know is that the gateway is there at all, which the provisional 1xx message accomplishes. What would be really nice is to have a timer that can be diffused as soon as a provisional reply arrives. I don't see how this can be accomplished with any of the existing timers. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 8

[Serusers] CDRTool error Table 'cdrtool.radacct200810' doesn't exist
by satish patel 13 Oct '08

13 Oct '08

Hi I have installed CDRTool and i got error on CDR search entry like this Table 'cdrtool.radacct200810' doesn't exist i have read on goolge about it and people tell the problem was if radacct200810 table not in radius database. i have also run rotatetable.php script to rotate database but i got error like this #php /var/www/CDRTool/scripts/SER/rotateTables.php rotateTable(radacct200810,200809,radacct200809) Error: cannot locate mysql creation file i am using 6.6.10 kindly suggest me.. $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

1 0

[Kamailio-Users] problem with relaying invite
by Asim Riaz 13 Oct '08

13 Oct '08

Hi, I am relaying invites from Kamilio(1.2) to a third proxy, i am receivng Server error occured in SIP messages, I am seeing these messages in log file; Asterisk--------->Kamilio----------------->third-proxy route{ openser.cfg if(method=="INVITE"){ route(4); exit; } route[4] { if(!t_relay("udp:third_proxy_1:5060") || !t_relay("udp:third_proxy_2:5060")) { end_media_session(); sl_reply_error(); } } DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found ERROR:tm:t_forward_nonack: no branch for forwarding ERROR:tm:w_t_relay: t_forward_nonack failed DEBUG:sl:sl_reply_error: error text is Server error occurred (1/SL) any idea ? Thanks Asim

3 7

[Kamailio-Users] what is the best solution NAT problem
by Ricky Gutierrez 13 Oct '08

13 Oct '08

Hi list, I am trying to solve my nat problem with openser 1.3.2 and rtpproxy 1.1, I have my openser with 2 net cards. I detail my scenario: (192.168.10.1) LAN-eth1- Server Openser eth0-WAN (192.168.1.64)<-> NAT <-> ADSL dyndns <-> Internet <-> ADSL <-> NAT <-> UAC All my external clients are also behind an ADSL with address private ip and my server openser, I don't have ip it public but register a domain with dyndns and it configures it in my router adsl, I have access from out to my server through dyndns. All my external clients configure them so that they use of address proxy the dyndns domain, my external clients register but when they call to another UAC that this behind the server openser is not audio, and if a UAC that this behind the server openser calls an external client the call doesn't arrive he fails. I have open the ports UDP 5060:5065, 10000:20000, 35000:65000 TCP: 5060 some idea of like I can solve this problem?... best regards all list rickygm ### LOG SIP ### U +1.744617 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000481 192.168.1.64:5060 -> 190.184.35.4:5063 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 232 P-hint: onreply_route|force_rtp_proxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35006 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=nortpproxy:yes # U +0.219110 190.184.35.4:5063 -> 192.168.1.64:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-547056da From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d" Contact: <sip:122@192.168.0.60:5063> User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000944 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.2 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-547056da From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d" Contact: <sip:122@190.184.35.4:5063;nat=yes> User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.140447 192.168.1.64:5060 -> 190.184.35.4:5063 OPTIONS sip:190.184.35.4:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0 From: sip:pinger@192.168.1.64;tag=f7508b75 To: sip:190.184.35.4:5063 Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64 CSeq: 1 OPTIONS Content-Length: 0 # U +0.164881 190.184.35.4:5063 -> 192.168.1.64:5060 SIP/2.0 404 Not Found To: sip:190.184.35.4:5063;tag=a04ea06caeb3256i3 From: sip:pinger@192.168.1.64;tag=f7508b75 Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64 CSeq: 1 OPTIONS Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0 Server: Linksys/SPA942-5.2.8 Content-Length: 0 # U +1.195009 192.168.10.28:5060 -> 192.168.10.1:5060 # U +0.639899 192.168.10.1:5060 -> 192.168.10.27:5060 OPTIONS sip:192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0 From: sip:pinger@192.168.1.64;tag=08508b75 To: sip:192.168.10.27:5060 Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1 CSeq: 1 OPTIONS Content-Length: 0 # U +0.001412 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0 From: sip:pinger@192.168.1.64;tag=08508b75 To: sip:192.168.10.27:5060;tag=b1e56101b3b09b53 Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1 CSeq: 1 OPTIONS User-Agent: Grandstream GXV3000 1.1.3.14 Contact: <sip:120@192.168.10.27:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U +3.393457 190.184.35.4:5063 -> 192.168.1.64:5060 BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-f7539b7 From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE Max-Forwards: 70 Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1" User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000768 192.168.10.1:5060 -> 192.168.10.30:5062 BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7 From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE Max-Forwards: 69 Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1" User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.055972 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0

3 3

Re: [Kamailio-Users] No audio both ways
by luzango mfupe 13 Oct '08

13 Oct '08

Hi Castillo,I did post my config file showing how i used the Mediaproxy. However i would like to get your opinion on how to use/configure it properly. cheers, Lu. > Message: 3 > Date: Sun, 12 Oct 2008 23:45:35 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] No audio both ways > To: users(a)lists.kamailio.org > Message-ID: <200810122345.35371.ibc(a)aliax.net> > Content-Type: text/plain; charset="utf-8" > > El Domingo, 12 de Octubre de 2008, luzango mfupe escribi?: > > I solved the problem by disabling use_mediaproxy() call and did put > nat=yes > > in the Asterisk side. > > This means that you didn't use MediaProxy correctly. If it's properly > configured and the proxy logic script calls use_mediaproxy() when needed, > then you don't need to enable the Comedia mode in Asterisk (nat=yes). > > -- > I?aki Baz Castillo > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 41, Issue 33 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 1

Re: [Kamailio-Users] No audio both ways
by luzango mfupe 12 Oct '08

12 Oct '08

Hi mates, I solved the problem by disabling use_mediaproxy() call and did put nat=yesin the Asterisk side. Many thanks to Daniel and David. Cheers, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 1

[Kamailio-Users] lcr module prefix_mode=1
by jh＠tutpro.com 12 Oct '08

12 Oct '08

by accident i sent this message to devel list when i intended to send it to users list. i'm thinking to speed up lcr load_gws() implementation by storing prefixes into hash table. it would allow load_gws() to execute (if necessary) in constant time independent of the number of prefixes. also, i'm planing to make the size of hash table controllable by module parameter. the side effect of this is that prefix_mode=1, where prefixes are regular expressions, would have to be dropped, because hashing on regular expressions makes no sense. something close to prefix_mode=1 could be achieved using dialplan module to store the regular expressions and return as attribute a gateway grp_id. that could then be used by new load_gws_from_group(grp_id) to load gws in the given group. any comments on this? is someone depending on prefix_mode=1? by the way, before starting this project, i took a look at opensips drouting module, but unfortunately it does not cover many features of lcr module that i use in my configurations and thus cannot be used to replace lcr module. -- juha

2 3

[Kamailio-Users] How can I validate a field HF of the Request with BD?
by Marcio 10 Oct '08

10 Oct '08

Hi all, I would like to know, how can i validate a field-hf(realm) of the request SIP, in case Register, with a column(realm/domain) of the table, which was created by me, in the mysql... will be that by avp ? or i will have of the create a module, for this...? Any advices welcome! cheers, Marcio

2 1

[Serusers] compiltation error when compiling avpops module in latest SER version from CVS
by Deepak 10 Oct '08

10 Oct '08

Hi, I am getting an error compiling avpops: make[1]: Entering directory `/opt/NEWSER/sip_router/modules/avpops' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev23-make"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c avpops.c -o avpops.o In file included from avpops_impl.h:41, from avpops_parse.h:38, from avpops.c:50: avpops_db.h:66: error: expected â)â before â*â token make[1]: *** [avpops.o] Error 1 make[1]: Leaving directory `/opt/NEWSER/sip_router/modules/avpops' make: *** [modules] Error 1 Any suggestions? Thanks

1 0

[Serusers] serctl moni problems
by psycho psycho 10 Oct '08

10 Oct '08

I haven' noticed but the whole error finally message by executing : sudo serctl moni is: [: 1074: ==: unexpected operator [: 1074: 0: unexpected operator _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

1 0

[Kamailio-Users] kamailio "failed to generate nonce"
by mayamatakeshi 10 Oct '08

10 Oct '08

Hello, I'm testing with kamailio 1.4 (just got revision 5050). I'm stressing it using a sipp scenario to verify which population of subscribers kamailio can handle. Using openser 1.3.3 I could easily keep thousands of subscribers online even when running on a VirtualMachine hosted on a desktop computer (last time I tested this it was about 8000 subscribers). But I'm running kamailio on a production machine and when I reach around 700 subscribers, kamailio starts to log this: Oct 9 17:44:20 ipx029 /usr/local/sbin/kamailio[7689]: ERROR:auth:build_auth_hf: no more nonces can be generated Oct 9 17:44:20 ipx029 /usr/local/sbin/kamailio[7689]: ERROR:auth:challenge: failed to generate nonce And the registration fails. Is anybody else seeing this? regards, takeshi

2 3

[Kamailio-Users] No audio both ways.
by luzango mfupe 10 Oct '08

10 Oct '08

Hi mates,I have this setup: Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port 5065) on the same Debian Box in Realtime with no NAT. Asterisk connects calls to the VoIP to PSTN provider. I am able to establish calls towards the PSTN side(Landline & Mobiles) but with no audio. I can hear the ringing tone but when the call connects and the conversation begin i hear nothing so as the Callee side. Below are my configs,the ngrep captured packets and codecs. #################################################################################### route[4] { # routing to the public network rewritehostport("xx.xxx.xxx.xx:5065"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } route[6] { # # -- NAT handling -- # if (isbflagset(6) || isbflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table #Gateway destinations are handled by regular expressions append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:0[1-9][0-9]+@.*") { if (is_user_in("credentials","local")) { strip(1); prefix("27"); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for local calls"); exit; }; }; if (uri=~"^sip:00[1-9][0-9]+@.*") { if (is_user_in("credentials","int")) { strip(2); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for international calls"); }; }; ################################################################################### This call was from the xlite softphone 1974 towards the Landline 0123825710. ################################################################################### U 2008/12/06 03:38:43.896057 196.212.209.18:46738 -> kamailio IP:5060 INVITE sip:0123825710@KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP 192.168.0.55: 46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward s: 70..Contact: <sip:1974@196.212.209.18:46738>..To: "0123825710"<sip:01238 25710@kamailio IP>..From: <sip:1974@kamailio IP>;tag=353dd217..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Allow: INVIT E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Cont ent-Type: application/sdp..User-Agent: X-Lite release 1014k stamp 47051..Co ntent-Length: 237....v=0..o=- 1 2 IN IP4 192.168.0.55..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3 101..a=fmtp :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 : ljiQYRpD NiMZsfdZ 192.168.0.55 60782..a=sendrecv.. U 2008/12/06 03:38:43.896350 Kamailio IP:5060 -> 196.212.209.18:46738 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.55:46 738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received =196.212.209.18..To: "0123825710"<sip:0123825710@kamailio IP>;tag=329cfea a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@kamailio IP>;tag=353dd217 ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Pr oxy-Authenticate: Digest realm="41.208.212.97", nonce="4939d8cfeb060ab14354 85eee811cdf644f759a2"..Content-Length: 0.... U 2008/12/06 03:38:44.086313 196.212.209.18:46738 -> kamailio IP:5060 ACK sip:0123825710@kamailio IP SIP/2.0..Via: SIP/2.0/UDP 192.168.0.55:467 38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To: "012382571 0"<sip:0123825710@kamailio IP>;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. .From: <sip:1974@41.208.212.97 <sip%3A1974(a)41.208.212.97>>;tag=353dd217..Call-ID: MGRiNzM0ZGYxZTk1ZDI3 ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0.... U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio IP;branch=z9hG4bKd78.7bd576 24.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738;received=1 96.212.209.18 ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673 8..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: <si p:1974@kamailio IP>;tag=353dd217..To: "0123825710"<sip:0123825710@41.208. 212.97>..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2 INV ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, RE FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: <sip:27123825710@41.2 08.212.97:5065>..Content-Length: 0.... U 2008/12/06 03:38:44.711225 70.42.72.49:5060 -> Asterisk IP:5065 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk IP:5065;branch=z9hG4b K22abec12;rport=5065..From: "1974" <sip:1974@Asterisk IP:5065>;tag=as6a7c b89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49>>..Call-ID: 1934f5d443abffe07 c59d0a42215b49c@41.208.212.97..CSeq: 102 INVITE..Server: OpenSER (1.3.2-not ls (i386/solaris))..Content-Length: 0.... U 2008/12/06 03:38:47.206445 70.42.72.49:5060 -> Asterisk IP:5065 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk IP:5065;branch=z9 hG4bK22abec12;rport=5065..From: "1974" <sip:1974@Asterisk IP:5065>;tag=as 6a7cb89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> >;tag=cba-1a6e-48ecbab6.. Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 INVITE.. Contact: <sip:1214650027123825710@70.42.72.138<sip%3A1214650027123825710(a)70.42.72.138>>..Date: Wed, 08 Oct 2008 13: 50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route: <sip:70.42.72.49 ;lr=on;fta g=as6a7cb89f>..Content-Type: application/sdp..Content-Length: 212....v=0..o =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN IP4 216.49 .201.22..t=0 0..m=audio 27852 RTP/AVP 0 101..a=ptime:20..a=rtpmap:0 PCMU/80 00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio IP;branch=z9hG4bK d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738; received=196.212.209.18 ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-; rport=46738..Record-Route: <sip:41.208.212.97 ;lr=on;ftag=353dd217;nat=yes>. .From: <sip:1974@kamilio IP>;tag=353dd217..To: "0123825710"<sip:01238257 10(a)41.208.212.97>;tag=as4377a96d..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Conta ct: <sip:27123825710@Asterisk IP:5065>..Content-Type: application/sdp..Co ntent-Length: 287....v=0..o=root 2664 2664 IN IP4 41.208.212.97..s=session. .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telepho ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=se ndrecv.. U 2008/12/06 03:38:47.207106 Kamailio IP:5060 -> 196.212.209.18:46738 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.55:46738 ;received= 196.212.209.18 ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467 38..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: <s ip:1974@Kamailio IP>;tag=353dd217..To: "0123825710"<sip:0123825710@kamilioIP>;tag=as4377a96d..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: <sip: 27123825710@41.208.212.97:5065>..Content-Type: application/sdp..Content-Len gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk IP..s=session..c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 PCMA/800 0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/ 8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. ######################################################################################## Sip.conf [general] context=from-trunk bindport=5065 autocreatepeer=yes bindaddr=xx.xxx.xxx.xx disallow=all ;allow=gsm ;allow=amr allow=alaw allow=ulaw allow=gsm ;allow=ilbc ;disallow=all ; ;useragent=Asterisk PBX dtmfmode = rfc2833 domain=xx.xxx.xxx.xx ; Add IP address as local domain [Provider] disallow=all canreinvite=no context=from-trunk allow=all ;allow=ulaw ;allow=gsm host=xx.xx.xx.xx insecure=port,invite type=peer ; we only want to call out, not be call$ dtmfmode=rfc2833 ######################################################################################### Here is my codecs 41*CLI> core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723 - - - - - - - - - - - - - - gsm - - 2 2 2 2 1 5 - 19 - 2 - 14 ulaw - 5 - 1 2 2 1 5 - 19 - 2 - 14 alaw - 5 1 - 2 2 1 5 - 19 - 2 - 14 g726aal2 - 5 2 2 - 2 1 5 - 19 - 1 - 14 adpcm - 5 2 2 2 - 1 5 - 19 - 2 - 14 slin - 4 1 1 1 1 - 4 - 18 - 1 - 13 lpc10 - 6 3 3 3 3 2 - - 20 - 3 - 15 g729 - - - - - - - - - - - - - - speex - 6 3 3 3 3 2 6 - - - 3 - 15 ilbc - - - - - - - - - - - - - - g726 - 5 2 2 1 2 1 5 - 19 - - - 14 g722 - - - - - - - - - - - - - - amr - 6 3 3 3 3 2 6 - 20 - 3 - - With best regards, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Register Kamailio with my service provider
by ingdavidcespedes＠cable.net.co 10 Oct '08

10 Oct '08

Hi! I'm trying to register my Sip Proxy Server (Kamailio) to my service provider, but I have to do it sending my user name and password. Right now I'm trying to make this work with Asterisk acting as a Gateway to my SIP SP, but there most be or should be a way to do it with kamailio, because based in SIP Connect Technical Recommendations the SPS of the client should register directly to the SPS of the Service Provider, without using any gateway. I know there is a way to make it with the UAC module, but only if the service provider match an IP address, not with user name and password. with best regards David Cespedes.

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