- sr-users - kamailio.org

[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.

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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.

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[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

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[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

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[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

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[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

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[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

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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: > 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

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[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

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[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......."

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

17 Oct '08

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

17 Oct '08

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[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

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[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

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[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......."

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[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

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[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

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[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

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[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

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[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

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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 >

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[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

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[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

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[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

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[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

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[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

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[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

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[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......."

3 2

[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.

3 2

[Serusers] How can I keep SER from changing reply Response Code from 503 to 500?
by Frank Durda IV 09 Oct '08

09 Oct '08

I have a situation where SER is gumming up the works. The problem is that I have a call where the INVITE fails because a PSTN gateway hasn't authorized the SIP invitee to call a given number, as intended. The PSTN switch is returning a "503 Service Unavailable" as is expected, but SER is then sending the SIP invitee a "500 Service Unavailable" instead of the original message, which is wrong in several ways. Here is the ngrep of what SER received from the PSTN switch (which is the desired answer in this situation): # U 2008/10/09 18:08:45.045015 10.133.0.2:5060 -> 10.133.90.6:5060 SIP/2.0 503 Service Unavailable. Call-ID: 14407805-3432564525-49719(a)sbc1.sometelco.com. CSeq: 1 INVITE. From: <sip:8019995678@209.x.x.x>;tag=3432564525-49731. To: <sip:9179891234@208.x.x.x:5060;user=phone>;tag=000a0285+1+4bae000a+de6c6831. Via: SIP/2.0/UDP 10.133.90.6;branch=z9hG4bKaa0f.96addb37.0. Via: SIP/2.0/UDP 209.x.x.x:5060;branch=z9hG4bKafec84c056d5d9ac97d27cc12d3dbb92. Server: DC-SIP/2.0. Content-Length: 0. Retry-After: 3. . That response code (503 Service Unavailable) is what should be sent back to the call originator, who can cope with it and redirect the call. However, here is the ngrep of what SER transmitted on to the invitee instead: # U 2008/10/09 18:08:45.050440 10.133.90.6:5060 -> 209.x.x.x:5060 SIP/2.0 500 Service Unavailable. To: <sip:9179891234@208.x.x.x:5060;user=phone>;tag=b11e68d9aa35b34c54f995457fe194f9-0069. From: <sip:8019995678@209.x.x.x>;tag=3432564525-49731. Call-ID: 14407805-3432564525-49719(a)sbc1.sometelco.com. CSeq: 1 INVITE. Via: SIP/2.0/UDP 209.x.x.x:5060;branch=z9hG4bKafec84c056d5d9ac97d27cc12d3dbb92. Server: Sip EXpress router (2.0.0-rc1 (x86_64/freebsd)). Content-Length: 0. . And the calling party receives the confusing "503 Service Unavailable". The onreply_route routine in use is a strip-down of what is found in the PSTN sample (page 54 of the SER Getting Started manual), and it makes no attempt to alter response codes: onreply_route["PSTN_REPLY"] { xlog("L_ERR","In onreply_route %rs %rr"); force_send_socket("10.133.90.6"); #emit on em1 fix_nated_contact("208.x.x.x"); /*Fixes Contact:*/ # Apply RTP proxy if necessary, but only for INVITE transactions # and 183 or 2xx replies if (@cseq.method != "INVITE") return; xlog("L_ERR","onreply_route[INVITE_REPLY] R1 %rs %rr"); if ((status =~ "(183)|2[0-9][0-9]") && search("^(Content-Type|c):.*application/sdp")) { xlog("L_ERR","In onreply_route FIXING PROXY"); force_rtp_proxy("e","208.x.x.x"); } return; } From debug messages (show below), the call to force_rtp_proxy() is not being performed, so in theory this routine is left with the 503 still intact. The xlog message shows the 503 value was still there on entry to onreply_route as well as near the end of the routine: Oct 9 18:08:45 ser1 /usr/local/sbin/ser[1820]: In onreply_route 503 Service Unavailable Oct 9 18:08:45 ser1 /usr/local/sbin/ser[1820]: onreply_route[INVITE_REPLY] R1 503 Service Unavailable The "FIXING PROXY' debug message was not reached, as expected. So what is causing: 503 Service Unavailable. to mysteriously turn into 500 Service Unavailable. which isn't even the correct RFC textual response for a 500 code. This is being seen in ser-2.0.0-rc1 and is completely repeatable. Ideas? Suggestions? Thanks in advance!

1 0

[Serusers] serctl moni problems
by psycho psycho 09 Oct '08

09 Oct '08

Hi, I have a problem with the serctl utility. Even though the ser process starts without any problems, when i am trying to use the monitor by executing serctl moni i get the following error [: 1074: 0: unexpected operator. All seems to work (making call etc) but monitor. If the number 1074 points to a line from the serctl then i attach is my serctl source (i don't know if that helps): 1073 cisco_restart) 1074 if [ "$#" -ne 2 ] ; then 1075 usage 1076 exit 1 1077 fi Any suggestions.? Thank you for your time. _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

1 0

[Kamailio-Users] Problem with call on hold
by Diego Zuaneti Arruda 09 Oct '08

09 Oct '08

Hi all, I'm having a problem when the user places the call on hold. When the user removes the call on hold, only one side hears the call. Does anyone know what is wrong? The dialog of the call is attached. Thanks. Diego.

2 2

Re: [Kamailio-Users] Users Digest, Vol 41, Issue 24
by luzango mfupe 09 Oct '08

09 Oct '08

Ok David, will give feedback soon.rgds, Lu. On Thu, Oct 9, 2008 at 11:53 AM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: No audio both ways. (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 9 Oct 2008 11:07:12 +0200 > From: "David Villasmil" <david.villasmil.work(a)gmail.com> > Subject: Re: [Kamailio-Users] No audio both ways. > Cc: users(a)lists.kamailio.org > Message-ID: > <9853f4ff0810090207p11d15bddt8dca8f57370c83c9(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Try adding a: > > nat=yes > > to the kamailio/openser peer definition and test > > dvg > > On Thu, Oct 9, 2008 at 10:58 AM, luzango mfupe <luzango.mfupe(a)gmail.com > >wrote: > > > > > 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@kamailioIP>;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@kamailioIP > >;tag=329cfea > > a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@kamailioIP > >;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@kamailioIP > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. > > .From: <sip:1974@41.208.212.97 <sip%3A1974(a)41.208.212.97> < > sip%3A1974(a)41.208.212.97 <sip%253A1974(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@AsteriskIP > :5065>;tag=as6a7c > > b89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(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@AsteriskIP > :5065>;tag=as > > 6a7cb89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(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> > <sip%3A1214650027123825710(a)70.42.72.138<sip%253A1214650027123825710(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@kamilio IP>;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......." > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > >

1 0

[Serusers] SIP MESSAGE BODY ENCRYPTION
by Adeyeye Oluwasegun Michael 09 Oct '08

09 Oct '08

Hi, I was wondering if the content or body of a SIP MESSAGE from a SIP client is always encrypted when a TLS-supported SIP proxy for the client registration and the likes. I know that S-MIME supported SIP stack could do that for me, but I was wondering if a TLS-supported proxy could do as well. Thanks, Michael.

2 1

[Kamailio-Users] SST module: Min-SE optional in request
by Schumann Sebastian 09 Oct '08

09 Oct '08

Dear all I might have noticed a bug in SST module: If I send an INVITE with Session-Expires: 300 Supported: timer I get rejected with 422 Session-Timer too small. If a Min-SE: x header with x > set min_se parameter included I do not get the message. This is IMHO wrong behavior, as Min-SE is optional in initial INVITE-requests sent by the phone acc. RFC 4028, table 1. Another thing that I noticed is the response to this message includes the Min-SE header (as it should), but has no linebreak after it (at least in kamailio branch 1.4). The message looks like MIN-SE:120Server: Kamailio (1.4.1-notls (i386/linux)) The phone call cannot be performed. At least one thing is IMHO wrong, as the phone cannot complete the call. It might be wrong the call gets rejected, or the answer might be possible like this, but then the phone cannot extract Min-SE due to wrong format. Any opinions on this? Sebastian

2 5

[Kamailio-Users] !!Urgent Need to setup SIP SERVER like SIP Account Provider as my own Server.Thanks for your very great Help.
by sam 09 Oct '08

09 Oct '08

HI , I want to be a SIP providers like broadvoice gimzo ,where can i get DID numbers ,do we need to have contact with telecommunication Department. Please give me a full details.When we use *kamailio *,then there is no need to get sip account and virtual numbers. -- Cheers K.S.Sampath Kumar

3 2

[Kamailio-Users] Kamailio and Asterisk trunking?
by John D 09 Oct '08

09 Oct '08

Not sure if I'm properly wording this, but I used to have two asterisk servers speaking to each other over SIP, configured via sip.conf. I now need a way to forward calls from my Asterisk instance to my Kamailio server so it can properly distribute the call. At this point however, I cannot find any documentation in order to properly have the two speak to each other. Would anyone be kind enough to point me in the right direction to establish that connection? Thanks.

3 2

[Kamailio-Users] Attended Transfer with asterisk
by ALAEDDINE abbech 08 Oct '08

08 Oct '08

Hello, Does anybody got a working configuration to make an "attended call transfer" with a call through an Asterisk gateway? Example: PSTN --> Asterisk --> SER --+-- A |<---transfer call pstn to B +-- B The call will come from the PSTN Network and will go through "A". A sets the call on "Hold" and calls "B". After A is connected with B, A hangup and B got the call from PSTN. This is not working at the moment. Attended call transfer only with OpenSER and only with SIP-Phones is no Problem. But if the is an Asterisk as PSTN-GW in the game it will not work.

3 4

[Kamailio-Users] ACK without matching transaction discarded
by Alex R.S.M 08 Oct '08

08 Oct '08

Hi, I am using openSER as a proxy for a simple SIP call (UAC <--> openSER <--> UAS). openSER receives ACK from callee but doesn't send it to caller. It seems openSER can not find a matching transaction, so it ignore and discard the message. if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } Here is the message header: Internet Protocol, Src: 10.0.0.174 (10.0.0.174), Dst: 10.0.0.160 (10.0.0.160) ... ACK sip:9134750000@10.0.0.183:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.174:5040;branch=z9hG4bK-12255-1-4 From: <sip:9133750000@10.0.0.174:5040>;tag=1 To: <sip:9134750000@10.0.0.183:5080>;tag=1 Call-ID: 1-12255(a)10.0.0.174 CSeq: 1 ACK Contact: sip:sipp@10.0.0.174:5040 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 Am I missing anything? Thanks, Alex _________________________________________________________________ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

2 2

[Kamailio-Users] Ping both UACs when a call starts
by Cesar Troncoso 08 Oct '08

08 Oct '08

Hello! When a call starts I would like OpenSER to ping (by sending OPTIONS or INFO request) both UAC´s just to know if they are still there or for some reason they crashed or lost connectivity. In case they don't response, the call should be finished. What is the best way to do this? Thanks in advance! Cesar

2 3

[Kamailio-Users] openser 1.3.3 domain module
by toly 07 Oct '08

07 Oct '08

Greetings, In domain module, command is_domain_local - using parameter_fixup where there is a call pkg_malloc, but nowhere function free_fixup is used. This will be memory leak, or I'm missing something. Thanks for the prompt reply, Toly -- View this message in context: http://www.nabble.com/openser-1.3.3-domain-module-tp19863638p19863638.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

[Kamailio-Users] LCR to Carrierroute with DNS/SRV Support
by Krom 07 Oct '08

07 Oct '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for URI manipulation prior to relaying to a cluster of session boarder controllers. As you know the DB table for LCR only supports integers in the gw table for ip_addr. I have been reading up on 1.2.x and 1.3.x to upgrade to a solution that would allow us to stay within the same call processing (script) design, but still support the new upstream requirement. I noticed the carrierroute table rewrite_host column is string. Could this be used to satisfy my needs of manipulating the URI for an IP where the carrier still uses a list of gateway IP's to try and use the DNS/SRV domain name where the carrier requires the use for SRV look-up? The session boarder controllers will conduct the SRV look-up after receiving the relayed message from Kamailio, it is just getting the domain name on the URI instead of IP address that I am looking to accomplish. Will the carrierroute module assist in this effort or is there another way that I am totally missing? Figured I would ask the gurus before following Alice down the rabbit hole. -- Thanks, -- Kyle

3 3

[Kamailio-Users] Can openser.cfg lead to pkg memory problem?
by mayamatakeshi 07 Oct '08

07 Oct '08

Hello, we have openser 1.3.3 running in production (current rev.: 4943). For 3 times in 50 days we had to restart openser to correct pkg memory problem. After some time logging messages like this: /openser.log:Aug 19 10:39:18 ipx022 /usr/local/sbin/openser[16991]: ERROR:core:new_credentials: no pkg memory left, openser will eventually run out of pkg memory and refuse all subsequent requests. We are trying to recreate this in our lab so that we can follow memory troubleshooting instructions at http://kamailio.net/dokuwiki/doku.php/troubleshooting:memory, but so far we were unable to do it even when generating millions of calls and registration transactions (we are using SIPp to generate normal call flows and even abnormal call flows detected when reading openser.log, like 'invalid cseq for aor', malformed SIP messages etc). And this is much more than in our production environment, with just 600 subscribers and about 2000 calls a day. The frequency the problem happens is increasing with the number of subscribers, so we are performing periodic restart of openser (actually, what we do is to switch over to the standby server). We already recompiled openser with pkg memory pool size set to 4MB so that this will not have to be done frequently. Since we cannot recreate this in our lab, we suspect there is a situation happening in production that might not be having been properly handled by openser.cfg. So my question is: would it be possible to an overlooked detail in openser.cfg to cause pkg memory problem? In case someone could take a look at it, here's our cfg file: ####### Global Parameters ######### debug=0 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/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:202.173.5.181:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "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" loadmodule "carrierroute.so" loadmodule "nathelper.so" loadmodule "dialog.so" loadmodule "snmpstats.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "lcr.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like 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 mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" loadmodule "uac.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_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", 1) # ----- 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) modparam("registrar", "min_expires", 30) modparam("registrar", "max_expires", 40) modparam("registrar", "default_expires", 35) # ----- 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", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "use_domain", 1) # ----- 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", "use_domain", 1) # ----- 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", "$avp(s:rpid)=rpid;$avp(s:blocked)=subscriber_status") # ----- 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") modparam("alias_db", "use_domain", 0) # ----- 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") # ----- carrieroute params ----- modparam("carrierroute", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("carrierroute", "config_source", "db") modparam("carrierroute", "use_domain", 1) # ----- NatHelper ----- #para versao a partir da versao 1.2 eh necessario esse paramtro para nao dar erro qdo usa a funcao "fix_nated_register();" modparam("nathelper|registrar", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- LCR ----- modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "dm_flag", 25) # ----- Dialog ---- modparam("dialog", "dlg_flag", 4) # ----- SnmpStat ----- modparam("snmpstats", "sipEntityType", "registrarServer") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 2000) modparam("snmpstats", "MsgQueueMajorThreshold", 5000) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_major_threshold", 750) modparam("snmpstats", "snmpgetPath","/usr/bin/") modparam("snmpstats", "snmpCommunity","public") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } ##nat route(2); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server #t_relay(); #exit; route(1); } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests setflag(4); #for dialog statistics # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) route(1); # t_relay(); # exit; } #t_check_trans(); if (is_method("PUBLISH|SUBSCRIBE|REFER|OPTIONS|MESSAGE")) { sl_send_reply("405", "Method not allowed"); exit; } # authenticate if from local subscriber (uncomment to enable auth) if (!(method=="REGISTER") && (!from_gw())) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; }else if (avp_check("$avp(s:blocked)", "eq/0")) { sl_send_reply("603","Subscriber disabled"); exit; }else if (avp_check("$avp(s:blocked)", "eq/1")) { sl_send_reply("603","Subscriber with outgoing blocked"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; }else if (avp_check("$avp(s:blocked)", "eq/0")) { sl_send_reply("403","Subscriber disabled"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); #if the call came from a known gateway it is not authenticated and we cannot use the function check_from() if (from_gw()) { route(4); }else if (!check_from()) {#if the check_from() returns false the call is not from a subscriber route(4); } else {#it is a subscriber, route using flip domain xlog("L_INFO", "routing using carrierroute $rm to $ru\n"); if (!cr_user_rewrite_uri("$fu", "flip")) { t_newtran(); t_reply("404", "No Route"); exit; } #replaces from by it's default DID uac_replace_from("sip:$avp(s:rpid)@$fd"); } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { xlog("L_INFO", "ROUTE_1 $rm to $ru\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')) { setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { xlog("L_INFO", "ROUTE_2 $rm to $ru\n"); if (method=="REGISTER") { fix_nated_register(); } else if (!from_gw()){ fix_nated_contact(); }; setflag(5); } route[3] { xlog("L_INFO", "ROUTE_3 $rm to $ru\n"); if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { xlog("L_INFO", "FORCE RTP w/ parameter.\n"); force_rtp_proxy("r"); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } route[4] { xlog("L_INFO", "uri does exist $rm to $ru \n"); if (alias_db_lookup("dbaliases")){ if (!lookup("location")) { switch ($retcode) { case -1: t_newtran(); t_reply("404", "Subscriber not online"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } }else{#check if did is blocked $rU = "(BLK)" + $rU; if (alias_db_lookup("dbaliases")){ sl_send_reply("403", "DID blocked"); exit; }else{# if it is not a valid DID nor a blocked DID tries to route it using peering domain if (!cr_rewrite_uri("peering", "call_id")) { t_newtran(); t_reply("404", "Peering Not Found"); exit; } } } } failure_route[1] { xlog("L_INFO", "FAILURE $rm to $ru\n"); if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { xlog("L_INFO", "ONREPLY_1 - Status $rs from $si $rm .\n"); if (is_method("INVITE")) { if ((isflagset(5)||isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (!from_gw()){ #if (isflagset(6)) { xlog("L_INFO", "ONREPLY_1 - ! from gw.\n"); fix_nated_contact(); } exit; } } Regards, takeshi

6 15

Re: [Kamailio-Users] Case Sensitive
by Daniel-Constantin Mierla 07 Oct '08

07 Oct '08

Please cc all the time to mailing lists, so everybody benefits of the outcome or can add comments/suggestions. Also, repeat my question, right now do you need it in some module or for string operations in the config file? Thank you, Daniel On 10/02/08 12:18, michel freiha wrote: > Dear Daniel, > > I prefer doing it in the config file...Is there any function that I > can add and it'll runn the server in case sensitive mode? > > Regards > > On Thu, Oct 2, 2008 at 11:42 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 10/02/08 11:10, michel freiha wrote: > > Dear All, > I would like to ask please about how to force kamailio to be > case sensitive regarding username? > > similar issue reported on the tracker: > http://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=1… > <http://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=1…> > > I guess it is same for you. Right now, do you need it in some > functions or in config file operations? There is need for special > functions for different comparisons, but also might be good to > have case sensitive operations in config file. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com <http://www.asipto.com/> > > -- Daniel-Constantin Mierla http://www.asipto.com

2 7

[Kamailio-Users] Dispatcher hash algorithms.
by Alex Balashov 06 Oct '08

06 Oct '08

I am curious, what exactly is the purpose and philosophy of the various hash algorithms in the dispatcher module? I am referring to the ones that allow the gateway in the route set to be determined through a hash of various SIP headers, such as the From URI, the To URI, the Call-ID GUID, etc. No guarantee of fair distribution - or in fact, any distribution that can be characterised by any describable pattern whatsoever - is implied by these algorithms. Additionally, it seems that without knowing (1) The exact hash algorithm in use; (2) The distribution that this hash algorithm would yield for a given set of possible values of these header fields, which in many cases are specifically intended to be pseudorandom (e.g. Call-ID), there is absolutely no way to determine, from any meaningfully deterministic perspective, which numerical entry in the route set these algorithms would computationally yield. So, I guess my question is: With no implied uniformity or weighting in the distribution whatsoever based on the incidental character of such values, what practical use does it serve to use any algorithms except round-robin or random? Is it expected that the user will plot the hash values against a log of given input strings to determine how the distribution will shape up? Is there some reason why the sort of profoundly lopsided distribution that may create might be desirable? Thanks, -- 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 11

[Kamailio-Users] CDRTools: Error instantiating db class:
by Tiago Rés 06 Oct '08

06 Oct '08

Hello all, When I login to CDRTool, it appears this error ("Error instantiating db class:") in the main page (CDRs). Any suggestion? Thanks! Tiago

1 0

[Serusers] lcr module and cancel
by Deepak 06 Oct '08

06 Oct '08

Hi, I have the following issue: I am using lcr module to select a gw from a set of gw's based upon R-URI prefixes. All is working fine. The issue is with a CANCEL that is sent by the UAC that originates the call before a dialog can be established. This implies that the CANCEL cannot be routed to the correct GW since SER see a CANCEL and simply returns "487 request terminated". How can I forward the CANCEL to the GW the the original INVITE was forwarded to? Any help appreciated.....

2 4

[Kamailio-Users] [offtopic] update vs delete/insert
by Daniel-Constantin Mierla 06 Oct '08

06 Oct '08

Hello, I am trying to get out you knowledge as DBA ... in some discussions in the past was brought out that in some cases doing a delete on primary key followed by a pure insert might be faster than one update. It might be dependent of DB server type and perhaps the question would have better answer in appropriate db-specific mailing lists, but I want to see if some of you has more experience/insights about this and is willing to share or got into same dilemma. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 4

[Kamailio-Users] Carrierroute module: preffix qty?
by Pablo Bertuzzi 06 Oct '08

06 Oct '08

Hi Forum, I want to use carrierroute to terminate 50,000 preffixes to a PSTN gw and block 100,000. So I created 50,000 records with gw ip as rewrite_host. However I only was able to start kamailio with no more than about 2,200 records in carrierroute mysql table. Is it possible to use carrierroute in such a way? Find below log with error: Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:bind_data_loader: use database as configuration source Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:load_route_data: name default, id 0, trees: 1 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: add carrier default Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: carrier default not found, add it Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_tree: tree default has internal id 0 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: created carrier tree: default, with id 0 and 1 trees Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_row: no private memory left Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_rows: error while converting row #2282 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_store_result: error while converting result Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:db_do_query: error while storing result Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:load_route_data: Failed to query database. Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:prepare_route_tree: could not load routing data Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:mod_init: could not prepare route tree Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:init_mod: failed to initialize module carrierroute Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:main: error while initializing modules

3 2

[Kamailio-Users] Developing Sh interface between the AS(OPENSER/OPENSIPS)and HSS
by suresh tattikota 06 Oct '08

06 Oct '08

hi all I am using the Openser(Opensips) as apllication server . i am developing Sh interface between the AS and HSS. For developing the Sh interface i am using the CDiameterpeer , here i am getiing memory and locking errors. And for using the CDiameterpeer as standalone library what steps i have to follow ... Thanks suresh :)

2 1

Re: [Kamailio-Users] nested redirect
by Daniel-Constantin Mierla 05 Oct '08

05 Oct '08

Hello, your main route block is something loke: route { if(is_method("INVITE")) { .... } exit; } So, apart of INVITE, every other sip request is dropped. See default config file to get a better understanding. kamailio/openser discards by default any sip request received if there is no action to forward for it. Cheers, Daniel Kent Pirlo wrote: > Attached is the full test config I am using.. I have gone back and forth trying a few things, but this is the latest.. If you have a another way to achieve the same result, I am open to change how I'm trying to do it. > > Thanks again for your help. > > Kent > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, September 23, 2008 5:50 AM > To: Kent Pirlo > Cc: users > Subject: Re: [Kamailio-Users] nested redirect > > Hello, > > sorry, you were right, there was a pcap attachment! The ACK looks ok. > Seems that ACK is quickly discarded by the script. Is the config you > pasted in your first email complete? I need to see it all the understand > how the ack get's routed and where could be dropped by the config. > > Cheers, > Daniel > > > > On 09/17/08 18:42, Kent Pirlo wrote: > >> There should have been a 20k attachment.. redirect.cap.. from tethereal. >> Let me know if you do not have it, and I will re-send it. >> >> kent >> >> -----Original Message----- >> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Sent: Wednesday, September 17, 2008 6:43 AM >> To: Kent Pirlo >> Cc: users >> Subject: Re: [Kamailio-Users] nested redirect >> >> Hello, >> >> what you sent is the debug from kamailio/openser, which is useful, but I >> would need the sip messages, you can use ngrep for that, do >> ngrep -d any -qt -W byline port 5060 >> >> on your openser/kamailio server and send the output. >> >> Thanks, >> Daniel >> >> >> On 09/17/08 02:22, Kent Pirlo wrote: >> >> >>> ... is my core switch, >>> ... is my openser/Kamailio redirect server >>> ... is a redirect server from a carrier of ours that returns a billing TN in contact header.. >>> >>> >>> Here is the debug of the incoming ACK from the core switch after being sent the 300... >>> >>> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: SIP Request: >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: method: <ACK> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: uri: <sip:...@...> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: version: <SIP/2.0> >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: flags=2 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to_param: tag=391854ebdd4d0c63cf50836cbb12191e-6708 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to: end of header reached, state=29 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to: display={}, >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: <To> [75]; >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: to body >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: cseq <CSeq>: <1> <ACK> >>> Sep 16 15:49:50 [6505] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKdce3ac34730f843a5a9b9c7002690f22>; state=16 >>> Sep 16 15:49:50 [6505] DBG:core:parse_via: end of header reached, state=5 >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: via found, flags=2 >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: this is the first via >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: After parse_msg... >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: preparing to run routing scripts... >>> Sep 16 15:49:50 [6505] DBG:sl:sl_filter_ACK: to late to be a local ACK! >>> Sep 16 15:49:50 [6505] DBG:core:destroy_avp_list: destroying list (nil) >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: cleaning up >>> >>> >>> >>> >>> -----Original Message----- >>> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Sent: Tuesday, September 16, 2008 5:10 AM >>> To: kentp(a)globalpops.com >>> Cc: users(a)lists.kamailio.org >>> Subject: Re: [Kamailio-Users] nested redirect >>> >>> Hello, >>> >>> can you post here the SIP messages? Will help to spot if there is >>> anything wrong with the ACK or something else in the signaling. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 09/13/08 04:11, kentp(a)globalpops.com wrote: >>> >>> >>> >>>> i have a simple redirect server working: >>>> >>>> Call comes from switch (A) to kamailio redirect server (B).. >>>> (B) runs perl script to determine route >>>> (B) sends Stateless 300 redirect reply to (A). >>>> (A) sends ACK to (B). >>>> >>>> Now, I need to add a nested redirect... >>>> >>>> call comes from sitch (A) to kamailio redirect server (B).. >>>> (B) then needs to send the call to third party proxy (C).. >>>> (C) will return 302 redirect.. >>>> (B) sends ACK to (C) >>>> (B) needs to read the contact header, then run perl script to determine route (using the contact value >>>> returned from (c)) >>>> (B) then sends 300 redirect back to (A). >>>> (A) sends ACK back to (B) >>>> >>>> >>>> so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending >>>> multiple 300's to (A): >>>> >>>> route { >>>> route(1); >>>> } >>>> >>>> route[1] { >>>> t_on_reply("1"); >>>> t_on_failure("2"); >>>> if (!t_relay("proxy.server.c:5060","0x02")) { >>>> sl_reply_error(); >>>> } >>>> onreply_route[1] { >>>> if(t_check_status("302")){ >>>> $var(lrn_from_bv) = $ct; >>>> } >>>> } >>>> failure_route[2] { >>>> >>>> if (t_check_status("3[0-9][0-9]")) { >>>> route(2); >>>> exit; >>>> } >>>> route[2] { >>>> if(perl_exec("callcheck")){ >>>> avp_pushto("$ru", "$avp(s:routes)/g"); >>>> t_reply("300","Go Here"); >>>> exit; >>>> } >>>> } >>>> >>>> >>>> >>>> -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] LCR module and CANCEL
by Deepak 04 Oct '08

04 Oct '08

Hi, I have the following issue: I am using lcr module to select a gw from a set of gw's based upon R-URI prefixes. All is working fine. The issue is with a CANCEL that is sent by the UAC that originates the call before a dialog can be established. This implies that the CANCEL cannot be routed to the correct GW since SER see a CANCEL and simply returns "487 request terminated". How can I forward the CANCEL to the GW the the original INVITE was forwarded to? Any help appreciated.....

1 0

[Kamailio-Users] Presence Server returns a 404 Not here msg
by Yoo Chan Jeon 03 Oct '08

03 Oct '08

Hi, I setup the presence server. My sip phone sends a PUBLISH msg. Th presence server sends a 200 msg. My another sip phone sends a SUBSCRIBE msg. The presence server returns 202 msg, and NOTIFY msg. These msgs does not include user in the contact header My sip phone sends a 200 msg I sends a second SUBSCRIBE msg to refresh it. This time my subscribe msg does not include the user in the request msg because the presence server did not include the user in the contact header. The presence server replied back with a 404 messge. So I looked at the codes, and add the user in the contact header when the presence server replys back to a client. But I stll have a same 404 msg. Do I miss something? I always appreciate your help. Thanks. Jeon

2 1

[Kamailio-Users] openser.cfg children=4
by Asim Riaz 03 Oct '08

03 Oct '08

Hi, my openser.cfg is configured for childern=4, but I am seeing child processes? Thanks Asim Riaz

3 6

Re: [Kamailio-Users] sermyadmin setup suse 11
by Ricky Gutierrez 03 Oct '08

03 Oct '08

I have looked for this inside the kamctl and I don't find it, I don't believe that it was that the export error, somebody that has installed sermyadmin in suse 11? regards .. rickygm Wednesday, October 1, 2008 12:24:06 PM, Iñaki Baz Castillo wrote: Subject: Re: [Kamailio-Users] sermyadmin setup suse 11 Please, don't send an image attachment, just copy the console output. This error is very common and is related to "phplib" field. Modify /etc/openserctl and set it to "no" (or something similar..). Look for the exact error in Google for more responses. --

2 1

[Kamailio-Users] Case Sensitive
by michel freiha 02 Oct '08

02 Oct '08

Dear All, I would like to ask please about how to force kamailio to be case sensitive regarding username? Regards

2 1

[Kamailio-Users] registrar module enhancements
by Daniel-Constantin Mierla 02 Oct '08

02 Oct '08

Hello, in the last time several enhancements were added to registrar module to help in dealing with user location records. This correlated with the new modue pv shall give lot of control over the destination set for calls. To get the latest devel version, check out the latest svn - svn co https://openser.svn.sourceforge.net/svnroot/openser/trunk kamailio The news in registrar module are: - ability to unregister all contacts for a SIP addres - there is a new function: unregister(table, uri) - ability to store and maintain single contact per user - the save() function accepts a new value for flags parameter (0x04) to restrict to one single record in location table - ability to access the attributes of all contacts for a user - there are two new functions: reg_fetch_contacts(table, uri, profile) and reg_free_contacts(profile) to make available/release the contacts for a SIP uri. After fetching the contacts, the attributes can be accessed via pseudo variable $ulc(profile=>attr) For more details see the README: http://www.kamailio.org/docs/modules/devel/registrar.html Here are couple of use cases: 1) only one contact per user save("location", "0x04"); Note that if you have couple of phones registering for same user, the one that sent last REGISTER message will be the only ringing. 2) check if a user is calling from a registered device and if not, deny the call if(!reg_fetch_contacts("location", "$fu", "caller")) { sl_send_reply("403", "Please register first"); exit; } $var(i) = 0; $var(found) = 0; $var(contact) = $(ct{nameaddr.uri}); while($var(found) == 0 && $var(i) < $(ulc(caller=>count))) { $if($var(contact)==$(ulc(caller=>addr)[$var(i)])) $var(found) = 1; else $var(i) = $var(i) + 1; } if($var(found) == 0) { sl_send_reply("403", "Please register first"); exit; } Testing and feedback is very much appreciated. For the future: - unregister() to get a $ulc(...) variable as parameter to unregister a specific contact - drop a branch via pv module Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 5

[Kamailio-Users] RLS(Resource List Server)
by Yoo Chan Jeon 02 Oct '08

02 Oct '08

Hi, I just want to know how RLS handles SUBSCRIBE messages without using XCAP. I assume that the server already has the list information. I want send the SUBSRIBE message with resource list format: SUBSCRIBE sip:list@domain . . supported:eventlist Accept:application/rlmi+xml Accept:multipart/related I added rls, presence, presence_xml, and pua modlues in the configuration file. I do not have any problems to run the server. Which database should I add list@doamin <list@doamin> to? How do I add the resource list? What kind of format do I need to add a list if I want to add joe@domain, bob@domain, tom@domain ? Do you have some kind of examples? Thanks you. Jeon

2 1

[Kamailio-Users] sermyadmin setup suse 11
by Ricky Gutierrez 01 Oct '08

01 Oct '08

Hi, I am trying to install sermyadmin 0.8 with the patch of security, but when I export the file openser.sql, it shows me this error: INSERT INTO `jsec_user_role_rel` (`id`, `version`, `role_id`, `user_id`) V' at line 1 if I try to export it again, it shows me an error message of having duplicated domain I erase the data of the domain chart, but same the same error ERROR 1062 (23000) at line 6: Duplicate entry '1' for key 1 God blesses, any help is welcome rickygm

2 1

[Serusers] two way routing
by Ivan Zweig 01 Oct '08

01 Oct '08

Hi, we are a small hosted VOIP company. We need a box that can take IN multiple origination providers and route the calls to the appropriate customer instance on our hosted platform. We use switchvox. We would also like to have 2-3 outbound routes to gain price efficiency and redundancy. I need someone that can implement this box and train us how to use it. you can email me to start a dialogue. We are looking for a consultant to help us with this.

1 0

[Kamailio-Users] About presence_dialoginfo - RR in figure 2
by Iñaki Baz Castillo 01 Oct '08

01 Oct '08

Hi, figure 2 of http://www.kamailio.org/docs/modules/devel/presence_dialoginfo.html shows the presence server sending the NOTIFY directly to the watcher. Is it possible to send the NOTIFY through the proxy? Well, since NOTIFY is in-dialog it's just needed that the proxy does RR in the initial SUBSCRIBE before forwarding it to presence server, am I right? Thanks. -- Iñaki Baz Castillo

3 3

[Kamailio-Users] new CDRTool release 6.6.9
by Adrian Georgescu 01 Oct '08

01 Oct '08

Hello, The software contains bug fixes for parallel prepaid calls. Changelog since 6.6.8 cdrtool (6.6.9) unstable; urgency=low * Purge old entries from prepaid_history table * Fixed wrong return to call control engine because sessionDoesNotExist variable was not initialized properly at each request * Print the call id in the syslog when calculating rate The software can be downloaded from: http://download.ag-projects.com/CDRTool/ For people running Debian testing or unstable on an i386 architecture there is an official public repository provided by AG Projects. To use it, 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 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 After that, run: apt-get update apt-get install cdrtool Regards, Adrian

1 0

[Kamailio-Users] LCR vs Carrierroute and SRV Support Inquirer
by Krom 30 Sep '08

30 Sep '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for RURI manipulation prior to relaying to a cluster of session boarder controllers. As you know the DB table for LCR only supports integers in the gw table for ip_addr. I have been reading up on 1.2.x and 1.3.x to upgrade to a solution that would allow us to stay within the same call processing (script) design, but still support the new upstream requirement. I noticed the carrierroute table rewrite_host column is string. Could this be used to satisfy my needs of manipulating the RURI for an IP where the carrier still uses a list of gateway IP's to try and use the DNS/SRV domain name where the carrier requires the use for SRV look-up? The session boarder controllers will conduct the SRV look-up after receiving the relayed message from Kamailio, it is just getting the domain name on the RURI instead of IP address that I am looking to accomplish. Will the (new to me) carrierroute module assist in this effort or is there another way that I am totally missing? Figured I would ask the gurus before following Alice down the rabbit hole. -- Thanks, -- Kyle

1 0

[Kamailio-Users] How to prevent - Same username used on multiple sip phones
by Ali Jawad 30 Sep '08

30 Sep '08

Hi On my openser I can logon with the same username on two different softphones. How can I prevent the same username from being used on multiple devices. With Regards

6 12

[Serusers] how-to AVP from radius load
by Samuel Muller 30 Sep '08

30 Sep '08

Hello, I've a single question about AVPs : how can I use them when loaded from radius_load_attr ? example : the 0123452000 is the caller, I want to use an acl to block him. => in my radius user config : 0123451010(a)sip.dot.com User-Password = "0000" User-Name = "0123451010(a)sip.dot.com", SER-UID = "0123451010(a)sip.dot.com", SER-DID = "1", SER-From-UID = "0123451010(a)sip.dot.com", SER-From-DID = "1", SER-Digest-Username = "0123451010", SER-Digest-Realm = "sip.dot.com", SER-Uri-User = "0123451010(a)sip.dot.com", SER-Attr = "acl:pstn", SER-Attr = "asserted-id:0123451010", SER-Attr = "phone:0123451010", SER-Attr = "lang:fr" => in the SER (v2.0.0) : if (!radius_load_attrs("$fu", "@f.uri")) { sl_reply("403", "Radius attributes not found"); exit; } if ( $acl == "pstn" ) { sl_reply("401", "Unauthorized"); exit; } I tried too " $fu.acl ", " set_attr function, ... it seems I can't use SER-Attr as a variable. in the debug (dump_attrs) : ... ser[28670]: r_INBOUND: => dump_attrs() ser[28670]: INFO: avp.c:542: class=GLOBAL ser[28670]: AVP["lang"]="en" ser[28670]: INFO: avp.c:552: track=FROM class=DOMAIN ser[28670]: AVP["did"]="1" ser[28670]: AVP["digest_realm"]="sip.dot.com" ser[28670]: INFO: avp.c:562: track=TO class=DOMAIN ser[28670]: AVP["did"]="1" ser[28670]: AVP["digest_realm"]="sip.dot.com" ser[28670]: INFO: avp.c:572: track=FROM class=USER ser[28670]: AVP["uid"]="0123451010(a)sip.dot.com" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451010" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451010" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451010" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451010" ser[28670]: INFO: avp.c:582: track=TO class=USER ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451005" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451005" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451005" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451005" ser[28670]: INFO: avp.c:592: track=FROM class=URI ser[28670]: AVP["uac_nat"]=1 ser[28670]: AVP["account"]="yes" ser[28670]: INFO: avp.c:602: track=TO class=URI ser[28670]: INFO: No AVP present ... If anyone has tips to help me, many thanks ! I'm sure there's something who replaces the old group_radius module, but which one or how to create SER-Attr $grp ? Many thanks, -- Samuel MULLER sml(a)720.fr

1 0

[Kamailio-Users] new CDRTool release 6.6.8
by Adrian Georgescu 30 Sep '08

30 Sep '08

Hello, The software includes critical bug fixes for parallel prepaid calls. The software can be downloaded from: http://download.ag-projects.com/CDRTool/ Changelog since 6.6.3 cdrtool (6.6.8) unstable; urgency=low * Log debit balance to prepaid_history table cdrtool (6.6.7) unstable; urgency=low * Do not debit balance for inexistent prepaid sessions cdrtool (6.6.5) unstable; urgency=low * Fixed purging of expired prepaid sessions cdrtool (6.6.4) unstable; urgency=low * Added links to Prepaid calls and quota in the main menu * Fixed showFooter() when exporting csv files from rating page * Fixed prepaid table mysql schema Regards, Adrian

1 0

[Kamailio-Users] kamailio 1.4.1 and mediaproxy 1.9.1
by toly 29 Sep '08

29 Sep '08

Greetings, I'm looking for the short answer, most likely no way. Thanks, Toly -- View this message in context: http://www.nabble.com/kamailio-1.4.1-and-mediaproxy-1.9.1-tp19728623p197286… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] Disable transfers (REFER) on Kamailio
by Diego Zuaneti Arruda 29 Sep '08

29 Sep '08

Hi all, It's possible to disable any transfers (blind or attended) on Kamailio ? Thanks. Diego.

2 2

Re: [Kamailio-Users] SDP ending with CRLF and a=nortpproxy:yes
by Schober Walter 29 Sep '08

29 Sep '08

I see: * [r4968] modules/nathelper/nathelper.c: - fix the calculation of CRLF length when adding norttptoxy string - patch by Alex Hermann (closes #2117700) And yes, it's that issue and therefore fixed in 1.4.1. > -----Ursprüngliche Nachricht----- > Von: Schober Walter > Gesendet: Samstag, 27. September 2008 14:13 > An: 'miconda(a)gmail.com'; Jerome Martin > Cc: users(a)lists.kamailio.org > Betreff: AW: [Kamailio-Users] SDP ending with CRLF and > a=nortpproxy:yes > > Hi! > > Same issue with 1.4.0. I'm migrating from 1.1.1 to 1.4.0, but > force_rtp_proxy() now adds a sdp line with a beginning 0x00, > so line looks like > 0000 00 61 3d 6e 6f 72 74 70 70 72 6f 78 79 3a 79 65 > .a=nortpproxy:ye > 0010 73 s > But no \r\n\r\n in front anymore in 1.4.0. > > Someone to direct me to the propper source file,pls? > > Walter > > > -----Ursprüngliche Nachricht----- > > Von: users-bounces(a)lists.kamailio.org > > [mailto:users-bounces@lists.kamailio.org] Im Auftrag von > > Daniel-Constantin Mierla > > Gesendet: Montag, 15. September 2008 14:23 > > An: Jerome Martin > > Cc: users(a)lists.kamailio.org > > Betreff: Re: [Kamailio-Users] SDP ending with CRLF and > > a=nortpproxy:yes > > > > Hi Jerome, > > > > On 09/15/08 13:43, Jerome Martin wrote: > > > Hi All, > > > > > > Just FYI, I had to disable the nortpproxy test here because > > it breaks > > > thomson ST2030 endpoints : with that line, they would just plain > > > ignore the messages containing the SDP ... > > was i with malformed SDP (empty lines inside SDP before the > a= line), > > or just the presence of the a=nortpproxy:yes line made the > ST2030 go > > mad? > > > > Cheers, > > Daniel > > > > > > > > Cheers, > > > > > > On Fri, 2008-09-12 at 18:12 +0300, Daniel-Constantin Mierla wrote: > > >> On 09/11/08 16:10, Klaus Darilion wrote: > > >> > A similar bug was also described in opensips. They tried > > to fix it > > >> > but the bug is still present. I wonder if this is > > something new or > > >> > if the bug is thre since the beginning of nathelper > but did not > > >> > occured until now > > >> > > > >> maybe this can be fixed by using the sdp parser available > > now in core. > > >> nathelper has quite limited sdp parsing capabilities now. > > the issue > > >> is to find the proper hook to add the lump with > > >> > > >> a=nortpproxy:yes\r\n" > > >> > > >> Cheers, > > >> Daniel > > >> > > >> > > >> > klaus > > >> > > > >> > Aymeric Moizard schrieb: > > >> > > > >> >> Hi, > > >> >> > > >> >> I just happen to find an issue with a software trying to call > > >> >> through my openser-1.3.1 > > >> >> > > >> >> After inserting "a=nortpproxy:yes\r\n", the message > coming out > > >> >> openser is: > > >> >> > > >> >> [normal sdp packet] > > >> >> a=fmtp:101 0-15\r\n > > >> >> \r\n > > >> >> \r\n > > >> >> "a=nortpproxy:yes\r\n" > > >> >> > > >> >> No matter who is responsible in that specific case (the SDP > > >> >> received by openser with several \r\n looks not > > compliant to me). > > >> >> > > >> >> I think openser should add "a=nortpproxy:yes" in > > different place > > >> >> in the SDP packet: NOT at the end of the SDP body. The > > "a=nortpproxy:yes" > > >> >> should be either put at the global attribute level or > > right below > > >> >> a m= line. > > >> >> > > >> >> This approach would solve such issue and would be much more > > >> >> adequate to the requirement: currently it's not very > > possible to > > >> >> relay an audio stream and not relay a video stream with the > > >> >> current approach. > > >> >> > > >> >> tks, > > >> >> Aymeric MOIZARD / ANTISIP > > >> >> amsip - http://www.antisip.com > > >> >> osip2 - http://www.osip.org > > >> >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > >> >> > > >> >> > > >> >> _______________________________________________ > > >> >> Users mailing list > > >> >> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > > >> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > >> >> > > >> > > > >> > _______________________________________________ > > >> > Users mailing list > > >> > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > > >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > >> > > > >> > > > >> > > >> > > > > > > *Jérôme Martin **| **LongPhone* > > > *Responsable Architecture Réseau* > > > 122, rue la Boetie | 75008 Paris > > > Tel : +33 (0)1 56 26 28 44 > > > Fax : +33 (0)1 56 26 28 45 > > > Mail : *jmartin*(a)longphone.fr > > > Web : www.longphone.com <http://www.longphone.com> > > > > > > > > > > > > ---------------------------------------------------------------------- > > > -- > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Daniel-Constantin Mierla > > http://www.asipto.com > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] CDRTool - stored procedures - not creating tables
by Graham Wooden 28 Sep '08

28 Sep '08

All, I have been having a heck of a time getting the radacct tables to be generated from the stored procedures, with the latest CDRTool. I have done and have reviewed them again the steps mentioned at: http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt Can anyone give any guidance. The call entries are showing up (for the most part) in the radacct table, but getting the main error upon viewing the CDR. ³Database error: Invalid SQL: select count(*) as records from radacct200809 where (AcctStartTime >= '2008-09-28 05:34' and AcctStartTime < '2008-09-28 23:55') MySQL error: 1146 (Table 'cdrtool.radacct200809' doesn't exist) 64Session halted.² I am using 1.4 Kamailio Thanks, -graham

1 0

Re: [Kamailio-Users] SDP ending with CRLF and a=nortpproxy:yes
by Schober Walter 27 Sep '08

27 Sep '08

Hi! Same issue with 1.4.0. I'm migrating from 1.1.1 to 1.4.0, but force_rtp_proxy() now adds a sdp line with a beginning 0x00, so line looks like 0000 00 61 3d 6e 6f 72 74 70 70 72 6f 78 79 3a 79 65 .a=nortpproxy:ye 0010 73 s But no \r\n\r\n in front anymore in 1.4.0. Someone to direct me to the propper source file,pls? Walter > -----Ursprüngliche Nachricht----- > Von: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] Im Auftrag von > Daniel-Constantin Mierla > Gesendet: Montag, 15. September 2008 14:23 > An: Jerome Martin > Cc: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] SDP ending with CRLF and > a=nortpproxy:yes > > Hi Jerome, > > On 09/15/08 13:43, Jerome Martin wrote: > > Hi All, > > > > Just FYI, I had to disable the nortpproxy test here because > it breaks > > thomson ST2030 endpoints : with that line, they would just plain > > ignore the messages containing the SDP ... > was i with malformed SDP (empty lines inside SDP before the > a= line), or just the presence of the a=nortpproxy:yes line > made the ST2030 go mad? > > Cheers, > Daniel > > > > > Cheers, > > > > On Fri, 2008-09-12 at 18:12 +0300, Daniel-Constantin Mierla wrote: > >> On 09/11/08 16:10, Klaus Darilion wrote: > >> > A similar bug was also described in opensips. They tried > to fix it > >> > but the bug is still present. I wonder if this is > something new or > >> > if the bug is thre since the beginning of nathelper but did not > >> > occured until now > >> > > >> maybe this can be fixed by using the sdp parser available > now in core. > >> nathelper has quite limited sdp parsing capabilities now. > the issue > >> is to find the proper hook to add the lump with > >> > >> a=nortpproxy:yes\r\n" > >> > >> Cheers, > >> Daniel > >> > >> > >> > klaus > >> > > >> > Aymeric Moizard schrieb: > >> > > >> >> Hi, > >> >> > >> >> I just happen to find an issue with a software trying to call > >> >> through my openser-1.3.1 > >> >> > >> >> After inserting "a=nortpproxy:yes\r\n", the message coming out > >> >> openser is: > >> >> > >> >> [normal sdp packet] > >> >> a=fmtp:101 0-15\r\n > >> >> \r\n > >> >> \r\n > >> >> "a=nortpproxy:yes\r\n" > >> >> > >> >> No matter who is responsible in that specific case (the SDP > >> >> received by openser with several \r\n looks not > compliant to me). > >> >> > >> >> I think openser should add "a=nortpproxy:yes" in > different place > >> >> in the SDP packet: NOT at the end of the SDP body. The > "a=nortpproxy:yes" > >> >> should be either put at the global attribute level or > right below > >> >> a m= line. > >> >> > >> >> This approach would solve such issue and would be much more > >> >> adequate to the requirement: currently it's not very > possible to > >> >> relay an audio stream and not relay a video stream with the > >> >> current approach. > >> >> > >> >> tks, > >> >> Aymeric MOIZARD / ANTISIP > >> >> amsip - http://www.antisip.com > >> >> osip2 - http://www.osip.org > >> >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ > >> >> > >> >> > >> >> _______________________________________________ > >> >> Users mailing list > >> >> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > >> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> >> > >> > > >> > _______________________________________________ > >> > Users mailing list > >> > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > >> > > >> > >> > > > > *Jérôme Martin **| **LongPhone* > > *Responsable Architecture Réseau* > > 122, rue la Boetie | 75008 Paris > > Tel : +33 (0)1 56 26 28 44 > > Fax : +33 (0)1 56 26 28 45 > > Mail : *jmartin*(a)longphone.fr > > Web : www.longphone.com <http://www.longphone.com> > > > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

1 0

[Kamailio-Users] towards kamailio v1.5.0
by Daniel-Constantin Mierla 26 Sep '08

26 Sep '08

Hello, as we just released 1.4.1 in the latest stable branch, I just want to give updates on the development version of the project. In less that two months since 1.4.x release series was branched, lot of new things happened - new modules and features, code refactoring and fixes. The outline of the new additions: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x A drafted version of the roadmap is available at: http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x I take this occasion to ask all developers to add there the new features already introduced and their plans for 1.5.0 in the above two pages. There are some features introduced on SVN that are not yet listed in Dokuwiki, features:new-in-1.5.x. So please, when you add some major improvement, add a short note about. As shortcut, those links are hosted on the website page: http://www.kamailio.org/mos/view/Roadmap/ For a deeper track of all changes you can browse the SVN repository at Source Forge. http://openser.svn.sourceforge.net/viewvc/openser/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

[Kamailio-Users] ip address of VIA header
by Alex R.S.M 26 Sep '08

26 Sep '08

Hi, Is there any function in openSER to return the ip address of VIA header? or return the VIA header? Thanks, alex _________________________________________________________________ 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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