- sr-users - kamailio.org

[OpenSER-Users] [CPL] subfield="tel" without "user=phone" parameter?
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

Hi, in order to match a callee URI as a telephone number (discarding punctuation and separator characters) I should user "tel" as "subfield" value, but this is only valid if "user=phone" parameters exists in the matching URI. RFC 3880 says: The "tel" subfield is defined to be the "user" part of the URI, with visual separators stripped, if the "user=phone" parameter is given to the URI, or if the server is otherwise configured to recognize the user part as a telephone number. Is there any way OpenSer CPL module to recognize "From" user part as telephone number even if no "user=phone" parameter exists? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [OpenSER-Users] [CPL] recurse="yes" doesn't work ? [SOLVED]
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

2007/12/24, Iñaki Baz Castillo <ibc(a)aliax.net>: > But in fact, the 302 arrives to the callee as if not "recurse" > parameter exists. Maybe I haven't understood the meaning of this > "recurse" parameter? or could it be a bug? Yes, a bug of mine XD For this feature working I must set a CPL parameter ("0 = dissable" by default): modparam("cpl-c","proxy_recurse",2) Now it works :) -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[OpenSER-Users] [CPL] recurse="yes" doesn't work ?
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

Hi, I try to set "recurse" parameter in order to automatically proxy the call to "Contact" URI present in a 302 response from called UAS: <lookup source="registration"> <success> <proxy timeout="8" recurse="yes"> </proxy> </success> </lookup> Now I set a redirection in the called UAS so it replies a 302. Reading RFC 3880 I assume that the proxy should create a branch to the 302 "Contact" URI instead of passing the 302 to the callee: 6.1. Proxy [...] The second optional parameter is "recurse", which can take two values, "yes" or "no". This specifies whether the server should automatically attempt to place further call attempts to telephony addresses in redirection responses that were returned from the initial server. [...] But in fact, the 302 arrives to the callee as if not "recurse" parameter exists. Maybe I haven't understood the meaning of this "recurse" parameter? or could it be a bug? Thanks for any explanation. Best regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[OpenSER-Users] Stopping branches from onreply_route?
by Helmut Kuper 24 Dec '07

24 Dec '07

Hi, is there a way to stop all branches of an initial INVITE from onreply_route, if a specific branch replies with code 486 ? regrads helmut

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[OpenSER-Users] SIP alternate port, cross connect with 5060 port
by Raju Abhyankar 24 Dec '07

24 Dec '07

Hello, I have setup openser (1.1.1-tls) to talk to our UA's on port 5060 and some UA's on port 4800 (as provider blocks 5060). I am using Mediaproxy (1.9.0). Currently the following works: 1. UA using 5060 makes call to UA using 4800: Phone bell rings and audio two way no issues. 2. UA using 4800 makes call to UA using 5060. Phone bell rings but no audio in either direction All UA's use RTP port 5004. What can I add to the following loose route section so that I can make "2" work above. My loose route section in openser is: ##################################################### listen= udp:XXX.XXX.XXX.XXX:5060 listen= udp:XXX.XXX.XXX.XXX:4800 # Loose route section if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) {(20) sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { (21)if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search ("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; ##################################################### Please do provide if there is a solution to this issue. Your inputs appreciated. regards, Roger

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[OpenSER-Users] T38 re invite not being sent to VoIP provider
by Mike O'Connor 23 Dec '07

23 Dec '07

Hi Guys I'm trying to get T38 working between our VoIP Provider and my epygi PBX. I have the epygi registered with openSER when a phone call is received from our provider openSER forwards the call to the epygi. When this unit detects the fax tone it does a re invite. (the re invite has the SDP indicating it want a T38) This re invite never gets to our VoIP provider. OpenSER is blocking the request. The invite would seem to be hitting our loose routing if (loose_route()) { xlog("L_NOTICE", "Running loose_route logic"); if ((is_method("INVITE|REFER") && ! has_totag())) { route(ROUTE_AUTHTEST); if (retcode == FALSE) return; }; if (method=="INVITE") { route(ROUTE_AUTHTEST); if (retcode == FALSE) return; }; append_hf("P-hint: rr-enforced\r\n"); route(ROUTE_DEFAULTHANDLER); return; }; route[ROUTE_DEFAULTHANDLER] { t_on_reply("REPLYROUTE"); if (!t_relay()) { sl_reply_error(); }; } Any ideas as to what I might have wrong ? Keep in mind that we are using OpenSER as a part of VoIP service to a small number of customers. Thanks Mike

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[OpenSER-Users] Saving Active Calls
by VoIP Forums www.Go4Calls.com 23 Dec '07

23 Dec '07

Hi Friends, How we can save Active Calls in openser mysql database? Please advise us i need to display it on our billing system. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[OpenSER-Users] Merry Christmas and a Happy New Year
by luzango.mfupe＠tuune.mobi 23 Dec '07

23 Dec '07

Mates, May you feel the Angels enfold you in their wings. May you always find serenity in the simple things. May the Light of Heaven shine upon your path, and bring you to the completion of your work in Peace and Joy and Grace. Wishing You All the Best this Christmas & a Happy new Year Welcome Year 2008 Tuune Mobile Bldg 6 Room 106,Staatsatillerie Rd FSATIE Tshwane University of Technology Pretoria West Republic of South Africa Website:WWW.Tuune.Mobi Tel:012-382-5953

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[OpenSER-Users] pv transformation does not work when Contact contains spaces
by Robert Dyck 23 Dec '07

23 Dec '07

When I included a pv transformation in my cfg script to extract the expires value from a Contact I found that it would not work if the Contact string contained spaces. I does not seem to affect registration. $(ct{param.value,expires}) The following patch in 1.3.0 works for me -- --- parser/parse_param.c-orig 2007-12-22 23:23:05.000000000 -0800 +++ parser/parse_param.c 2007-12-22 23:23:38.000000000 -0800 @@ -281,7 +281,7 @@ while(_s->len) { switch(_s->s[0]) { - case ' ': + //case ' ': case '\t': case '\r': case '\n': An example of a message that fails Contact expires transformation -- Session Initiation Protocol Request-Line: REGISTER sip:proxy01.sipphone.com SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.4:5061;branch=z9hG4bK-1849b3b3 From: Gateway <sip:17476316719@proxy01.sipphone.com>;tag=a0bda46794e0291o1 To: Gateway <sip:1234567@proxy01.sipphone.com> Call-ID: 6d0c06a5-8d11d464(a)192.168.1.4 CSeq: 51635 REGISTER Max-Forwards: 70 Authorization: Digest username="1234567",realm="proxy01.sipphone.com",nonce="476e0e7866cafbf20434684456644a3d2022c85f",uri="sip:proxy01.sipphone.com",algorithm=MD5,response="57d8e4aea67f9bb45ced0394fe54d659" Contact: Gateway <sip:1234567@192.168.1.4:5061>;expires=3600 User-Agent: Linksys/SPA3000-3.1.10(GWd) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura

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[Serusers] (more details) Solution for the NAT IP changed?
by Gonzalo J. Sambucaro 23 Dec '07

23 Dec '07

Folcks, The problem =========== supose that there is an established rtp connection between two endpoints and the media proxy is in the middle doing the relay of the rtp streams, let´s say (MP = mediaproxy) EP A <--------->[NAT with IP1]<-----------> MP <--------------------> EP B EP A sends rtp to MP_IP:MP_PORT passing through the NAT box. EP B sends rtp to MP_IP:MP_PORT without passing through a NAT box. The MP know that the caller = NAT_IP1:NAT_PORT1, and the called = EP_B_IP:EP_B_PORT Now, supose that the NAT box change their PUBLIC IP from IP1 to IP2, so this escenary EP A <------->[NAT with IP1]<---------> MP <-------------> EP B will change to this EP A <------->[NAT with IP2]<---------> MP <--------------> EP B so the MP should detect that change of IPs and continue relaying the rtp streams but now to IP2:PORT2 instead of IP1:PORT1. Well, that was the situation y have experienced. The solution ============ To fix this, I was thinking this solution: 1) When the first rtp packet of a source arrives, save the SSRC field in the MP. - Save the SSRC of the caller. - Save the SSRC of the called. 2) If arrives a rtp packet with unknown source IP but with the same SSRC field of some of the two streams, updates the binding (with the new IP detected) between the caller and the MP or between the called and the MP according to the field SSRC previously saved. Note: SSRC (RFC 3550 RTP), (from the rfc: "The SSRC identifier carried in the RTP header and in various fields of RTCP packets is a random 32-bit number that is required to be globally unique within an RTP session ") What do you think about the solution? Regards, -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

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[Serusers] Forking request - destination in branches?
by Tomasz Zieleniewski 22 Dec '07

22 Dec '07

Hi, I have following issue. All my outgoing requests are forwarded through boundary proxy by setting the destination before t_relay() my problem is when there is another branch created. This second branch is routed in the standard SIP way directly to destination in request URI. How can I force also branches to go through my boundary proxy. Can I somehow check in branch_route that this branch is not the first one (not first message) but the second and and change the destination and make t_relay()? Thank You for any feedback Regards Tomasz

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[Serusers] SER + asterisk integration config
by satish patel 22 Dec '07

22 Dec '07

Dear all anybody have done this job integration SER with Asterisk ????? $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- 5, 50, 500, 5000 - Store N number of mails in your inbox. Click here.

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[OpenSER-Users] UAC gives
by Raju Abhyankar 22 Dec '07

22 Dec '07

Hello, I am using openser-1.1 and have the standard script from onsip. I am using Mediaproxy for NAT traversal. Most of my clients register for days with no issues. A few clients after say a day of operation suddenly start giving getting de-registered. I can see the ATA or UAC trying to register to openser but openser gives a reply back to the UAC saying "401 Unauthorized". The UAC's which give this issue are behind standard home grade wireless routers (netgear/linksys type). Any idea why this may happen. Thanks in advance Roger

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[OpenSER-Users] Any Sip Phone support examples/web_im/click_to_dial.php? Mine failed, UA says "481 Call Leg/Transaction Does Not Exist"
by CHEN XUEQIN 21 Dec '07

21 Dec '07

Hi all: I know that the example uses REFER sip method and follow such process: 1. prepare two SIP softphone(eyebeam 1.5.16.1 ) , register to openser 2. user enter Caller and Callee's SIP Address 3. Web Server receive http request, then construct FIFO command, using t_uac_dlg MI command, build Refer SIP message with From and To header using user input 4. t_uac_dlg is invoked, send sip REFER message to Caller Sip softphone, softphone return 481 Call Leg/Transaction Does Not Exist, so the call can't be setup. I have seen the notice"/Unfortunately, this example does not work. The reason is use of REFER for third-party call-control has not been standardized due to resistance of proponents of B2BUA use (which is somewhat bloated and not always operational)". But I still wonder if any softphone support this example? My openser version: 1.3.0 Thanks. Regards, Chen Xueqin /

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[OpenSER-Users] Merry Christmas and a Happy New Year 2008!
by Daniel-Constantin Mierla 21 Dec '07

21 Dec '07

I wish Merry Christmas and a Happy New Year 2008 to everybody helping with the project, from code to feedback and reports. It was a full year of activities, with two major releases, lot of enhancements, fixes, and participations to events around the world. I think OpenSER fulfills most of the requirements its users need and it will keep being focused on robustness, performance, flexibility and new enhancements. Looking forward to a 2008 of successes. Cheers, Daniel

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[OpenSER-Users] OpenSER SST module for session timer
by Pat wang 21 Dec '07

21 Dec '07

Happy holidays! Not sure what happened to my previous email the format is kind of messy. Sorry about that. I am playing with SST module for session timer and have some questions that I would like someone can help me out. 1. First question is the 422 message is not sent immediately after receiving INVITE with SE header smaller than default 1800 sec. My OpenSER script is very strait forward as the example in the module document: if (method=="INVITE") { if (sstCheckMin("1")) { blah blah.... exit; } } Debug on OpenSER shows the result returns -1 when an INVITE contains SE and Min-SE timer at 90. It is until the INVITE hits lookup("location") it replies a 422. 2. Right after OpenSER sends a 422 it proxies the INVITE to the called party at that same time. So the call flow looks like: Caller OpenSER Callee INVITE (SE 90)=======> <======= 422 ==== ======INVITE===> === INVITE (SE 1800)==> 3. The 422 Session Timer Too Small message put MIN-SE and Server headers in one line which looks like: MIN-SE: 1800Server: OpenSER........... Has anyone tried SST module before and how did it work for you? Many thanks for you help. Best regards, Patrick _________________________________________________________________ The best games are on Xbox 360. Click here for a special offer on an Xbox 360 Console. http://www.xbox.com/en-US/hardware/wheretobuy/

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[OpenSER-Users] (no subject)
by Pat wang 21 Dec '07

21 Dec '07

Happy holidays! I am playing with SST module for session timer and have some questions that I would like someone can help me out. 1. First question is the 422 message is not sent immediately after receiving INVITE with SE header smaller than default 1800 sec. My OpenSER script is very strait forward as the example in the module document: if (method=="INVITE") { if (sstCheckMin("1")) { blah blah.... exit; }}Debug on OpenSER shows the result returns -1 when an INVITE contains SE and Min-SE timer at 90. It is until the INVITE hits lookup("location") it replies a 422. 2. Right after OpenSER sends a 422 it proxies the INVITE to the called party at that same time. So the call flow looks like: Caller OpenSER Callee INVITE (SE 90)=======> <======= 422 ==== ======INVITE===> === INVITE (SE 1800)==> 3. The 422 Session Timer Too Small message put MIN-SE and Server headers in one line which looks like: MIN-SE: 1800Server: OpenSER........... Has anyone tried SST module before and how did it work for you? Many thanks for you help. Best regards,Patrick _________________________________________________________________ Share life as it happens with the new Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_122007

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[OpenSER-Users] Newbie
by ravimondi＠wlink.com.np 21 Dec '07

21 Dec '07

Hello all, I would like to know where could the materials for openser configurations be found for a newbie like me? Thank you

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[Serusers] fifo question
by Strassburger, Patrick 21 Dec '07

21 Dec '07

Hi all, I am struggling arround since hours about fifo. I get all the time a syntax error. Tired several things like chmod, cr /tmp/ser_fifo as a file or directory, created a /temp/ser_fifo. Changed ser.cfg to fifo="/temp/ser_fifo" fifo_mode=0666 with / without "" does not help all the time a syntax error. Can someone explain - what fifo is (file or folder) - would gues it is a file (first in first out)? - does fifo needs to be created before? I changed /temp/ or /tmp/ to 666 but this does not help either. starting ser shows me [root@ser /]# ser -f /etc/ser/ser.cfg -c 0(8574) parse error (23,1-5): syntax error 0(8574) parse error (23,1-5): ERROR: bad config file (2 errors) wherin line 23 is the one with fifo. Kind regards Patrick # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=<ip adress> # put your real IP here port=5060 alias=sip.mycompany.com children=4 server_signature=yes dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=yes reply_to_via=no uid="nobody" gid="nobody" mhomed=0 fifo="/temp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900.

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Re: [OpenSER-Users] Ignore CPL script
by Jesus Rodriguez 20 Dec '07

20 Dec '07

Hello Martin, Please, keep always the list in copy as more people can be interested in the replies. > I am beginner in this. I have this as project to school. In which > part of > config file I have to write > cpl_run_script("incoming","force_stateful") ? You can have something like: if (is_method("INVITE") { # run incoming script if ( !cpl_run_script("incoming","force_stateful") ) { # script execution failed sl_send_reply("500","CPL script execution failed"); exit; }; }; Regards. Saludos JesusR. >> Hi Martin, >> >> >> El 20/12/2007, a las 16:28, Martin Kardi escribió: >> >> [...] >> >>> and this is easy cpl script: >>> >>> <?xml version="1.0" encoding="UTF-8"?> >>> >>> <cpl> >>> <incoming> >>> <reject status="400" reason="Busy" /> >>> </incoming> >>> </cpl> >>> >>> Upload of this cpl script to DB is correct, record in table "cpl" is >>> created >>> for "user" but when I am calling to this "user", openser ignores >>> this script >>> and phone is ringing. >>> I don't know what is problem, could somebody help me? >> >> >> You need to call the cpl_run_script() function to execute the >> uploaded >> cpl script. Something like: >> >> cpl_run_script("incoming","force_stateful") >> >> Regards. >> >> Saludos >> JesusR. >> >> ------------------------------------ >> Jesus Rodriguez >> VozTelecom Sistemas, S.L. >> jesusr(a)voztele.com >> http://www.voztele.com >> Tel. 902360305 >> ------------------------------------- >> >> >> >> >> > > > Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

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[Serusers] Solution for the NAT IP changed?
by Gonzalo J. Sambucaro 20 Dec '07

20 Dec '07

Hello folks, I was Working in the NAT IP changed solution, my idea is to change media proxy server (MPS) in a such way when in a established connection and receive a RTP package from unknown IP source (because a NAT server changed their IP), using de field SSRC (RFC 3550 RTP) updates the binding (with the new IP detected) between the caller and the MPS or between the called and the MPS according to the field SSRC previously saved. What do you think about my solution? Regards, Gonzalo Sambucaro. -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

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[OpenSER-Users] OpenSER 1.2.3 patch release is out!
by Bogdan-Andrei Iancu 20 Dec '07

20 Dec '07

Hi everybody, The third patch release of 1.2.x series is out as version 1.2.3. It preserves full compatibility with 1.2 family regarding the configuration file. OpenSER 1.2.3 brings all kind of fixes (mainly from the 1.3 version). No new features were added. The 1.2.3 release includes: - small fixes (compile warnings, porting across UNIX platform errors - bug fixing - mi_xmlrpc update to work with latest 1.0 libxmlrpc-c3 releases - functionality fixes (Route header handling in local sequential requests) - small improvements - more self - sanity checks, more verbose error codes in script (like t_relay() in TM). Tomorrow I will start uploading the tarballs on the web sites. Best regards, Bogdan

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[OpenSER-Users] Ignore CPL script
by Martin Kardi 20 Dec '07

20 Dec '07

Hi, I have problem with CPL. This is my config file for openser: debug=3 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/build/buildd/openser-1.1.0/debian/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/build/buildd/openser-1.1.0/debian/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/build/buildd/openser-1.1.0/debian/openser/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" #cpl modul -- dopisali sme tu my !!!!!! loadmodule "/usr/lib/openser/modules/cpl-c.so" # Uncomment this if you want digest authentication #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #---CPL-C Parameter modparam("cpl-c","db_url","mysql://openser:ser@localhost/openser") modparam("cpl-c","cpl_table","cpl") modparam("cpl-c","cpl_dtd_file","/etc/openser/cpl-06.dtd") modparam("cpl-c","log_dir","/var/log/openser/cpl") #modparam("cpl-c","proxy_recurse",2) #modparam("cpl-c","proxy_route",0) #modparam("cpl-c","nat_flag",6) modparam("cpl-c","lookup_domain","location") #cpl_run_script("incoming","force_statefull") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; cpl_process_register(); save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; and this is easy cpl script: <?xml version="1.0" encoding="UTF-8"?> <cpl> <incoming> <reject status="400" reason="Busy" /> </incoming> </cpl> Upload of this cpl script to DB is correct, record in table "cpl" is created for "user" but when I am calling to this "user", openser ignores this script and phone is ringing. I don't know what is problem, could somebody help me? thanks Martin -- View this message in context: http://www.nabble.com/Ignore-CPL-script-tp14438450p14438450.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Need help troubleshooting flags in onreply route - v.1.2.2
by Dan-Cristian Bogos 20 Dec '07

20 Dec '07

Guys, I would need your help in troubleshooting some pretty serious issue for me. Scenario: relaying an INVITE from openser to a softswitch and recognizing the flags (global and branch ones) in onreply. Everything seems to work fine with relaying the requests, but when the answers are matched in onreply, the flags are gone. Bellow is the log configuration which gives the output used by me to troubleshoot in logs: in route: xlog("First branch: $br, all branches: $bR, flags set: $mf , branch flags: $bf"); xlog("Relaying to: $avp(s:route), message: $mb"); xlog("OUT to Softswitch"); in onreply route: xlog ("ENTERING ON REPLY ROUTE"); xlog("First branch: $br, all branches: $bR, flags set: $mf , branch flags: $bf"); xlog("In onreply, message: $mb"); Please see the output bellow: Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: First branch: <null>, all branches: <null>, flags set: 130 , branch flags: 1024 Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: Relaying to: pstn1, message: INVITE sip:INT_NUMBER_MASKED@sip.itsyscom.com SIP/2.0^M Via: SIP/2.0/UDP 10.10.11.10:7091;rport;branch=z9hG4bK2127028508^M From: <sip:dan@sip.itsyscom.com>;tag=2128765320^M To: <sip:INT_NUMBER_MASKED@sip.itsyscom.com>^M Call-ID: 886593274(a)10.10.11.10^M CSeq: 21 INVITE^M Contact: <sip:dan@10.10.11.10:7091>^M Proxy-Authorization: Digest username="dan", realm="sip.itsyscom.com", nonce="476a40226a7cfaa9a11de5b6e7d6742d45ec6bd4", uri="sip:INT_NUMBER_MASKED@sip.itsyscom.com", response="08872e4a06f011ea7d6f947f1874a882", algorithm=MD5^M Max-Forwards: 4^M User-Agent: Linphone-1.5.1/eXosip^M Subject: Phone call^M Expires: 120^M Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE^M Content-Type: application/sdp^M Content-Length: 310^M ^M v=0^M o=dan 123456 654321 IN IP4 10.10.11.10^M s=A conversation^M c=IN IP4 10.10.11.10^M t=0 0^M m=audio 7078 RTP/AVP 3 0 8 111 110 101^M a=rtpmap:3 GSM/8000/1^M a=rtpmap:0 PCMU/8000/1^M a=rtp Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: OUT to Softswitch Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: ENTERING ON REPLY ROUTE Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: First branch: <null>, all branches: <null>, flags set: 0 , branch flags: 0 Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: In onreply, message: SIP/2.0 100 Trying^M Via: SIP/2.0/UDP IP_MASKED;branch=z9hG4bK9221.886b3817.0;received=IP_MASKED;rport=5060^M Via: SIP/2.0/UDP 10.10.11.10:7091;received=ORIGINATOR_IP;rport=8026;branch=z9hG4bK2127028508^M Record-Route: <sip:IP_MASKED;lr=on;ftag=2128765320;nat=yes>^M From: <sip:dan@sip.itsyscom.com>;tag=2128765320^M To: <sip:INT_NUMBER_MASKED@sip.itsyscom.com>^M Call-ID: 886593274(a)10.10.11.10^M CSeq: 21 INVITE^M Server: GlobalE SoftSWITCH/1.3.0.^M Content-Length: 0^M ^M Dec 20 10:07:53 ellie /usr/local/sbin/openser[15296]: ENTERING ON REPLY ROUTE Dec 20 10:07:53 ellie /usr/local/sbin/openser[15296]: First branch: <null>, all branches: <null>, flags set: 0 , branch flags: 0 Dec 20 10:07:53 ellie /usr/local/sbin/openser[15296]: In onreply, message: SIP/2.0 180 Ringing^M Via: SIP/2.0/UDP IP_MASKED;branch=z9hG4bK9221.886b3817.0;received=IP_MASKED;rport=5060^M Via: SIP/2.0/UDP 10.10.11.10:7091;received=ORIGINATOR_IP;rport=8026;branch=z9hG4bK2127028508^M Record-Route: <sip:IP_MASKED;lr=on;ftag=2128765320;nat=yes>^M From: <sip:dan@sip.itsyscom.com>;tag=2128765320^M To: <sip:INT_NUMBER_MASKED@sip.itsyscom.com>;tag=975999519^M Call-ID: 886593274(a)10.10.11.10^M CSeq: 21 INVITE^M Server: GlobalE SoftSWITCH/1.3.0.^M Contact: <sip:INT_NUMBER_MASKED@IP_MASKED:5070>^M Allow: ACK, INVITE, BYE, CANCEL, OPTIONS^M Content-Type: application/sdp^M Content-Length: 187^M ^M v=0^M o=yate 1198145273 1198145273 IN IP4 208.50.127.243^M s=SIP Call^M c=IN IP4 208.50.127.243^M t=0 0^M m=audio 23166 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M Can any of you help me troubleshooting this? I get the feeling that there is no transaction matched by openser for the replies. Version I am using (compiled from sources, with some params modified in configure.h): openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:3368M @(#) $Id: main.c 3173 2007-11-20 08:26:35Z bogdan_iancu $ Thxs in advance for any help. DanB

2 2

[Serusers] SER To Address
by Arun Kumar 20 Dec '07

20 Dec '07

Hi, When I send calls to PSTN the From filed is my user@SER IP is correct but in To Filed I want Number@ProviderIP but SER send Number@SERIP ? Thanks, Arun

1 0

Re: [OpenSER-Users] Openser and Nokia E61
by David Villasmil Govea 19 Dec '07

19 Dec '07

Ravi or Juha, if you've connected E61 succesfully, could you post the parameters here? I just can't... Thanks a lot David

3 2

Re: [OpenSER-Users] What is the difference?
by David Villasmil Govea 19 Dec '07

19 Dec '07

> > > El Wednesday 19 December 2007 13:20:07 I?aki Baz Castillo escribi?: > > El Wednesday 19 December 2007 13:15:35 David Villasmil Govea escribi?: > > > What's the difference between > > > > > > setbflag and setflag > > > ?? > > > > http://www.openser.org/dokuwiki/doku.php/utils:flags > > Note that if you use setbflag into a "route" you set the bflag for ALL the > branches (there could be more than one branch because pararell forking and > so). > > If you want to set bflag just for a branch you should use "setbflag" > into "branch_route". > > And NEVER do "isbflagset" into a "route" since the result can be > unexpectable > (if a branch has that bflag active and other one not active, "isbflagset" > in "route" could return 1 or 0 depending on which branch examinates > before). > > I ask, because i had this cfg file that was working perfectly with nat/rtpproxy. I replace it with a new cfg that was using bflag instead of flag. client-rtpproxy-client rtps stopped working. When i replace those bflags with regular flags it started working again!... Also, as i read there, "per branch" means each location... talking about location, is there a reason a client would have 98 almost exactly alike locations? but for the id, they're the same... weird. And i started seeing lots of "max branch exceeded" on the syslog. We do not limit branches because a client could have his softphone and ata logged in... thanks a lot

2 1

[OpenSER-Users] Problems with presence
by Jon Farmer 19 Dec '07

19 Dec '07

Hi I am trying to get presence working and getting the following error. Can anyone suggest what it wrong? I am using openser 1.2.2 notls and xlite and eyebeam for testing. Dec 19 17:11:47 esosser openser[22325]: PRESENCE: generate_ETag: etag= a.1198083751.22325.1 / 20 Dec 19 17:11:47 esosser openser[22325]: PRESENCE: handle_publish: No E-Tag and no body present Dec 19 17:11:47 esosser openser[22325]: PRESENCE: handle_publish: ERROR occured Regards Jon __________________________________________________________ Sent from Yahoo! Mail - a smarter inbox http://uk.mail.yahoo.com

1 0

[OpenSER-Users] SoftPhone with CPL script upload support
by Martin Kardi 19 Dec '07

19 Dec '07

Hi, I try find some softphone with uploading CPL scripts support? Please advise me somebody. Thanks Martin -- View this message in context: http://www.nabble.com/SoftPhone-with-CPL-script-upload-support-tp14419452p1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[OpenSER-Users] ERROR:avp:add_avp: no more shm mem
by Marc LEURENT 19 Dec '07

19 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning Dan and Good Morning everybody, I've got Freeradius 1.1.7 + (python module enable + rlm_digest patches) OpenSER 1.2.2 + (Radius ACC enable) CDRTool 5.3.2 and I don't understand these outputs... OpenSER is crashing after receiving Authorization from Freeradius... Any idea?? Freeradius sent AVP to OpenSER: SIP-AVP = "callMaxDur:2957" SIP-AVP = "credit:10.0000" SIP-AVP = "authentic#1" And openser is trying to parse it but the there is a problem... 2(30175) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 2(30175) DEBUG:auth_radius:extract_avp: string is <ur:2957> 2(30175) DEBUG:auth_radius:extract_avp: AVP name is <ur> 2(30175) DEBUG:auth_radius:extract_avp: AVP val is <2957> 2(30175) ERROR:avp:add_avp: no more shm mem 2(30175) ERROR:auth_radius:generate_avps: Unable to create a new AVP 10(30190) ERROR: receive_fd: EOF on 6 Thanks ### Small Freeradius output ### A2 = INVITE:sip:0677832974@sip.wifirst.net H(A1) = b82eaab7b4eaab2db812d3a2695f0e22 H(A2) = 83134a8c6b9ea1a20de012abf16e87f5 KD = b82eaab7b4eaab2db812d3a2695f0e22:47679476280e04ea50ded04b24f7d3f75068eea7:83134a8c6b9ea1a20de012abf16e87f5 EXPECTED 40ed14d07aa16035e67c21886a802379 RECEIVED 40ed14d07aa16035e67c21886a802379 modcall[authenticate]: module "digest" returns ok for request 71 modcall: leaving group authenticate (returns ok) for request 71 Login OK: [mleurent(a)sip.wifirst.net/<no User-Password attribute>] (from client private port 5060) Processing the post-auth section of radiusd.conf modcall: entering group post-auth for request 71 rlm_sql (sql): Processing sql_postauth radius_xlat: 'mleurent(a)sip.wifirst.net' rlm_sql (sql): sql_set_user escaped user --> 'mleurent(a)sip.wifirst.net' radius_xlat: 'INSERT into radpostauth (id, user, pass, reply, date) values ('', 'mleurent(a)sip.wifirst.net', 'Chap-Password', 'Access-Accept', NOW())' rlm_sql (sql) in sql_postauth: query is INSERT into radpostauth (id, user, pass, reply, date) values ('', 'mleurent(a)sip.wifirst.net', 'Chap-Password', 'Access-Accept', NOW()) rlm_sql (sql): Reserving sql socket id: 2 rlm_sql (sql): Released sql socket id: 2 modcall[post-auth]: module "sql" returns ok for request 71 modcall: leaving group post-auth (returns ok) for request 71 Sending Access-Accept of id 180 to 172.24.0.8 port 47861 SIP-AVP = "callMaxDur:2957" SIP-AVP = "credit:10.0000" SIP-AVP = "authentic#1" Finished request 71 Going to the next request - --- Walking the entire request list --- Waking up in 6 seconds... ### Full OpenSER output ### 2(30175) SIP Request: 2(30175) method: <INVITE> 2(30175) uri: <sip:0677832974@sip.wifirst.net> 2(30175) version: <SIP/2.0> 2(30175) parse_headers: flags=2 2(30175) get_hdr_field: cseq <CSeq>: <2> <INVITE> 2(30175) Found param type 232, <branch> = <z9hG4bK3c2eaf3d-baab-dc11-90b2-ed571fe4ba74>; state=6 2(30175) Found param type 235, <rport> = <n/a>; state=17 2(30175) end of header reached, state=5 2(30175) parse_headers: Via found, flags=2 2(30175) parse_headers: this is the first via 2(30175) After parse_msg... 2(30175) preparing to run routing scripts... 2(30175) parse_headers: flags=100 2(30175) DEBUG:parse_to:end of header reached, state=10 2(30175) DBUG:parse_to: display={}, ruri={sip:0677832974@sip.wifirst.net} 2(30175) DEBUG: get_hdr_field: <To> [34]; uri=[sip:0677832974@sip.wifirst.net] 2(30175) DEBUG: to body [<sip:0677832974@sip.wifirst.net> ] 2(30175) DEBUG: get_hdr_body : content_length=235 2(30175) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(30175) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 50 2(30175) !!!! STARTING ROUTE 2: TEST NAT 2(30175) !!!!!!!!! YAHH NAT DETECTED - FIXING CONTACT!!!!!!!!!!!!! 2(30175) parse_headers: flags=80 2(30175) DEBUG: add_param: tag=2eb08b3d-baab-dc11-90b2-ed571fe4ba74 2(30175) DEBUG:parse_to:end of header reached, state=29 2(30175) DBUG:parse_to: display={"Marc LEURENT"}, ruri={sip:mleurent@sip.wifirst.net} 2(30175) parse_headers: flags=200 2(30175) found end of header 2(30175) find_first_route: No Route headers found 2(30175) loose_route: There is no Route HF 2(30175) grep_sock_info - checking if host==us: 15==12 && [sip.wifirst.net] == [213.91.4.219] 2(30175) grep_sock_info - checking if port 5060 matches port 5060 2(30175) JUSTE BEFORE SETTING FLAGS FOR RADIUS 2(30175) DEBUG: has_totag: no totag 2(30175) I AM SETTING THE FLAGS FOR RADIUS 2(30175) SETTING FLAGS 1 FOR RADIUS 2(30175) SETTING FLAGS 2 FOR RADIUS 2(30175) grep_sock_info - checking if host==us: 15==12 && [sip.wifirst.net] == [213.91.4.219] 2(30175) grep_sock_info - checking if port 5060 matches port 5060 2(30175) grep_sock_info - checking if host==us: 15==12 && [sip.wifirst.net] == [213.91.4.219] 2(30175) grep_sock_info - checking if port 5060 matches port 5060 2(30175) check_nonce(): comparing [47679476280e04ea50ded04b24f7d3f75068eea7] and [47679476280e04ea50ded04b24f7d3f75068eea7] 2(30175) DEBUG:auth_radius:radius_authorize_sterman: Success 2(30175) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 2(30175) DEBUG:auth_radius:extract_avp: string is <ur:2957> 2(30175) DEBUG:auth_radius:extract_avp: AVP name is <ur> 2(30175) DEBUG:auth_radius:extract_avp: AVP val is <2957> 2(30175) ERROR:avp:add_avp: no more shm mem 2(30175) ERROR:auth_radius:generate_avps: Unable to create a new AVP 10(30190) ERROR: receive_fd: EOF on 6 10(30190) DBG: handle_ser_child: dead child 2, pid 30175 (shutting down?) 10(30190) DBG: io_watch_del (0x5b5c20, 6, -1, 0x0) fd_no=13 called 0(30173) child process 30175 exited by a signal 11 0(30173) core was generated 0(30173) INFO: terminating due to SIGCHLD 4(30177) INFO: signal 15 received 4(30177) Memory status (pkg): 4(30177) fm_status (0x5f6d80): 4(30177) heap size= 1048576 4(30177) used= 77952, used+overhead=122432, free=926144 4(30177) max used (+overhead)= 129560 4(30177) dumping free list: 4(30177) hash = 1 fragments no.: 4, unused: 0 bucket size: 8 - 8 (first 8) 4(30177) hash = 2 fragments no.: 6, unused: 0 bucket size: 16 - 16 (first 16) 4(30177) hash = 3 fragments no.: 4, unused: 0 bucket size: 24 - 24 (first 24) 4(30177) hash = 4 fragments no.: 1, unused: 0 bucket size: 32 - 32 (first 32) 4(30177) hash = 5 fragments no.: 1, unused: 0 bucket size: 40 - 40 (first 40) 4(30177) hash = 6 fragments no.: 4, unused: 0 bucket size: 48 - 48 (first 48) 4(30177) hash = 7 fragments no.: 25, unused: 0 bucket size: 56 - 56 (first 56) 4(30177) hash = 8 fragments no.: 6, unused: 0 bucket size: 64 - 64 (first 64) 4(30177) hash = 9 fragments no.: 14, unused: 0 bucket size: 72 - 72 (first 72) 4(30177) hash = 10 fragments no.: 2, unused: 0 bucket size: 80 - 80 (first 80) 4(30177) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 4(30177) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 4(30177) hash = 28 fragments no.: 1, unused: 0 bucket size: 224 - 224 (first 224) 4(30177) hash = 29 fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) 4(30177) hash = 52 fragments no.: 2, unused: 0 bucket size: 416 - 416 (first 416) 4(30177) hash = 62 fragments no.: 1, unused: 0 bucket size: 496 - 496 (first 496) 4(30177) hash = 64 fragments no.: 1, unused: 0 bucket size: 512 - 512 (first 512) 4(30177) hash = 65 fragments no.: 1, unused: 0 bucket size: 520 - 520 (first 520) 4(30177) hash = 146 fragments no.: 1, unused: 0 bucket size: 1168 - 1168 (first 1168) 4(30177) hash = 176 fragments no.: 1, unused: 0 bucket size: 1408 - 1408 (first 1408) 4(30177) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 917296) 4(30177) TOTAL: 79 free fragments = 926336 free bytes 4(30177) ----------------------------- 5(30178) INFO: signal 15 received 5(30178) Memory status (pkg): 5(30178) fm_status (0x5f6d80): 5(30178) heap size= 1048576 5(30178) used= 77776, used+overhead=121040, free=927536 5(30178) max used (+overhead)= 121504 5(30178) dumping free list: 5(30178) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 5(30178) hash = 41 fragments no.: 1, unused: 0 bucket size: 328 - 328 (first 328) 5(30178) hash = 55 fragments no.: 1, unused: 0 bucket size: 440 - 440 (first 440) 5(30178) hash = 56 fragments no.: 1, unused: 0 bucket size: 448 - 448 (first 448) 5(30178) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 926496) 5(30178) TOTAL: 5 free fragments = 927728 free bytes 5(30178) ----------------------------- 1(30174) INFO: signal 15 received 1(30174) Memory status (pkg): 1(30174) fm_status (0x5f6d80): 1(30174) heap size= 1048576 3(30176) INFO: signal 15 received 3(30176) Memory status (pkg): 3(30176) fm_status (0x5f6d80): 3(30176) heap size= 1048576 3(30176) used= 77936, used+overhead=122400, free=926176 3(30176) max used (+overhead)= 129576 3(30176) dumping free list: 3(30176) hash = 1 fragments no.: 6, unused: 0 bucket size: 8 - 8 (first 8) 3(30176) hash = 2 fragments no.: 4, unused: 0 bucket size: 16 - 16 (first 16) 3(30176) hash = 3 fragments no.: 1, unused: 0 bucket size: 24 - 24 (first 24) 3(30176) hash = 4 fragments no.: 2, unused: 0 bucket size: 32 - 32 (first 32) 3(30176) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) 3(30176) hash = 7 fragments no.: 24, unused: 0 bucket size: 56 - 56 (first 56) 3(30176) hash = 8 fragments no.: 3, unused: 0 bucket size: 64 - 64 (first 64) 3(30176) hash = 9 fragments no.: 16, unused: 0 bucket size: 72 - 72 (first 72) 3(30176) hash = 10 fragments no.: 4, unused: 0 bucket size: 80 - 80 (first 80) 1(30174) used= 77992, used+overhead=122472, free=926104 1(30174) max used (+overhead)= 129696 1(30174) dumping free list: 1(30174) hash = 1 fragments no.: 6, unused: 0 bucket size: 8 - 8 (first 8) 1(30174) hash = 2 fragments no.: 5, unused: 0 bucket size: 16 - 16 (first 16) 1(30174) hash = 3 fragments no.: 3, unused: 0 bucket size: 24 - 24 (first 24) 1(30174) hash = 6 fragments no.: 6, unused: 0 bucket size: 48 - 48 (first 48) 1(30174) hash = 7 fragments no.: 29, unused: 0 bucket size: 56 - 56 (first 56) 1(30174) hash = 8 fragments no.: 3, unused: 0 bucket size: 64 - 64 (first 64) 1(30174) hash = 9 fragments no.: 16, unused: 0 bucket size: 72 - 72 (first 72) 1(30174) hash = 11 fragments no.: 2, unused: 0 bucket size: 88 - 88 (first 88) 1(30174) hash = 19 fragments no.: 1, unused: 0 bucket size: 152 - 152 (first 152) 1(30174) hash = 28 fragments no.: 1, unused: 0 bucket size: 224 - 224 (first 224) 1(30174) hash = 29 fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) 1(30174) hash = 52 fragments no.: 2, unused: 0 bucket size: 416 - 416 (first 416) 1(30174) hash = 96 fragments no.: 1, unused: 0 bucket size: 768 - 768 (first 768) 1(30174) hash = 150 fragments no.: 1, unused: 0 bucket size: 1200 - 1200 (first 1200) 1(30174) hash = 176 fragments no.: 1, unused: 0 bucket size: 1408 - 1408 (first 1408) 1(30174) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 917848) 1(30174) TOTAL: 79 free fragments = 926296 free bytes 1(30174) ----------------------------- 3(30176) hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) 3(30176) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) 3(30176) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 3(30176) hash = 28 fragments no.: 2, unused: 0 bucket size: 224 - 224 (first 224) 3(30176) hash = 29 fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) 3(30176) hash = 52 fragments no.: 2, unused: 0 bucket size: 416 - 416 (first 416) 3(30176) hash = 58 fragments no.: 1, unused: 0 bucket size: 464 - 464 (first 464) 3(30176) hash = 59 fragments no.: 1, unused: 0 bucket size: 472 - 472 (first 472) 3(30176) hash = 102 fragments no.: 1, unused: 0 bucket size: 816 - 816 (first 816) 3(30176) hash = 122 fragments no.: 1, unused: 0 bucket size: 976 - 976 (first 976) 3(30176) hash = 128 fragments no.: 1, unused: 0 bucket size: 1024 - 1024 (first 1024) 3(30176) hash = 176 fragments no.: 1, unused: 0 bucket size: 1408 - 1408 (first 1408) 3(30176) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 916008) 3(30176) TOTAL: 78 free fragments = 926368 free bytes 3(30176) ----------------------------- 10(30190) INFO: signal 15 received 10(30190) Memory status (pkg): 10(30190) fm_status (0x5f6d80): 10(30190) heap size= 1048576 10(30190) used= 127600, used+overhead=170816, free=877760 10(30190) max used (+overhead)= 170816 10(30190) dumping free list: 10(30190) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 877952) 10(30190) TOTAL: 1 free fragments = 877952 free bytes 10(30190) ----------------------------- 9(30188) INFO: signal 15 received 9(30188) Memory status (pkg): 9(30188) fm_status (0x5f6d80): 9(30188) heap size= 1048576 9(30188) used= 135792, used+overhead=179024, free=869552 9(30188) max used (+overhead)= 179024 9(30188) dumping free list: 9(30188) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 9(30188) TOTAL: 1 free fragments = 869744 free bytes 9(30188) ----------------------------- 6(30181) INFO: signal 15 received 6(30181) Memory status (pkg): 6(30181) fm_status (0x5f6d80): 6(30181) heap size= 1048576 6(30181) used= 135792, used+overhead=179024, free=869552 6(30181) max used (+overhead)= 179024 6(30181) dumping free list: 6(30181) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 6(30181) TOTAL: 1 free fragments = 869744 free bytes 6(30181) 8(30185) INFO: signal 15 received 8(30185) Memory status (pkg): 8(30185) fm_status (0x5f6d80): 8(30185) heap size= 1048576 8(30185) used= 135792, used+overhead=179024, free=869552 8(30185) max used (+overhead)= 179024 8(30185) dumping free list: 8(30185) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 8(30185) TOTAL: 1 free fragments = 869744 free bytes 8(30185) ----------------------------- - ----------------------------- 7(30182) INFO: signal 15 received 7(30182) Memory status (pkg): 7(30182) fm_status (0x5f6d80): 7(30182) heap size= 1048576 7(30182) used= 135792, used+overhead=179024, free=869552 7(30182) max used (+overhead)= 179024 7(30182) dumping free list: 7(30182) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 7(30182) TOTAL: 1 free fragments = 869744 free bytes 7(30182) ----------------------------- 1(30174) INFO: signal 15 received 1(30174) Memory status (pkg): 1(30174) fm_status (0x5f6d80): 1(30174) heap size= 1048576 1(30174) used= 85664, used+overhead=128704, free=919872 1(30174) max used (+overhead)= 128704 1(30174) dumping free list: 1(30174) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 1(30174) hash = 42 fragments no.: 1, unused: 0 bucket size: 336 - 336 (first 336) 1(30174) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 919640) 1(30174) TOTAL: 3 free fragments = 920064 free bytes 1(30174) ----------------------------- 0(30173) pool_remove: Connection still kept in the pool 0(30173) pool_remove: Connection still kept in the pool 0(30173) XLOG: destroy module ... 0(30173) pool_remove: Removing connection from the pool 0(30173) pool_remove: Connection still kept in the pool 0(30173) INFO:mi_fifo:mi_destroy: seems that fifo child is already dead! 0(30173) pool_remove: Removing connection from the pool 0(30173) DEBUG: tm_shutdown : start 0(30173) DEBUG: unlink_timer_lists : emptying DELETE list 0(30173) DEBUG: tm_shutdown : emptying hash table 0(30173) DEBUG: tm_shutdown : releasing timers 0(30173) DEBUG: tm_shutdown : removing semaphores 0(30173) DEBUG: tm_shutdown : destroying tmcb lists 0(30173) DEBUG: tm_shutdown : done 0(30173) Memory status (pkg): 0(30173) fm_status (0x5f6d80): 0(30173) heap size= 1048576 0(30173) used= 56464, used+overhead=99648, free=948928 0(30173) max used (+overhead)= 112768 0(30173) dumping free list: 0(30173) hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) 0(30173) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) 0(30173) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) 0(30173) hash = 4 fragments no.: 33, unused: 0 bucket size: 32 - 32 (first 32) 0(30173) hash = 6 fragments no.: 9, unused: 0 bucket size: 48 - 48 (first 48) 0(30173) hash = 7 fragments no.: 5, unused: 0 bucket size: 56 - 56 (first 56) 0(30173) hash = 8 fragments no.: 1, unused: 0 bucket size: 64 - 64 (first 64) 0(30173) hash = 9 fragments no.: 2, unused: 0 bucket size: 72 - 72 (first 72) 0(30173) hash = 12 fragments no.: 9, unused: 0 bucket size: 96 - 96 (first 96) 0(30173) hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) 0(30173) hash = 14 fragments no.: 3, unused: 0 bucket size: 112 - 112 (first 112) 0(30173) hash = 21 fragments no.: 2, unused: 0 bucket size: 168 - 168 (first 168) 0(30173) hash = 156 fragments no.: 2, unused: 0 bucket size: 1248 - 1248 (first 1248) 0(30173) hash = 842 fragments no.: 1, unused: 0 bucket size: 6736 - 6736 (first 6736) 0(30173) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 936000) 0(30173) TOTAL: 86 free fragments = 949120 free bytes 0(30173) ----------------------------- 0(30173) Memory status (shm): 0(30173) fm_status (0x2ab19f9d0000): 0(30173) heap size= 33554432 0(30173) used= 144, used+overhead=36720, free=33517712 0(30173) max used (+overhead)= 2746832 0(30173) dumping free list: 0(30173) hash = 1 fragments no.: 56, unused: 0 bucket size: 8 - 8 (first 8) 0(30173) hash = 2 fragments no.: 12, unused: 0 bucket size: 16 - 16 (first 16) 0(30173) hash = 3 fragments no.: 9, unused: 0 bucket size: 24 - 24 (first 24) 0(30173) hash = 4 fragments no.: 10, unused: 0 bucket size: 32 - 32 (first 32) 0(30173) hash = 5 fragments no.: 3, unused: 0 bucket size: 40 - 40 (first 40) 0(30173) hash = 6 fragments no.: 4, unused: 0 bucket size: 48 - 48 (first 48) 0(30173) hash = 7 fragments no.: 47, unused: 0 bucket size: 56 - 56 (first 56) 0(30173) hash = 8 fragments no.: 3, unused: 0 bucket size: 64 - 64 (first 64) 0(30173) hash = 9 fragments no.: 6, unused: 0 bucket size: 72 - 72 (first 72) 0(30173) hash = 10 fragments no.: 2, unused: 0 bucket size: 80 - 80 (first 80) 0(30173) hash = 12 fragments no.: 1, unused: 0 bucket size: 96 - 96 (first 96) 0(30173) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 0(30173) hash = 17 fragments no.: 1, unused: 0 bucket size: 136 - 136 (first 136) 0(30173) hash = 21 fragments no.: 5, unused: 0 bucket size: 168 - 168 (first 168) 0(30173) hash = 22 fragments no.: 1, unused: 0 bucket size: 176 - 176 (first 176) 0(30173) hash = 35 fragments no.: 1, unused: 0 bucket size: 280 - 280 (first 280) 0(30173) hash = 76 fragments no.: 1, unused: 0 bucket size: 608 - 608 (first 608) 0(30173) hash = 104 fragments no.: 1, unused: 0 bucket size: 832 - 832 (first 832) 0(30173) hash = 110 fragments no.: 1, unused: 0 bucket size: 880 - 880 (first 880) 0(30173) hash = 112 fragments no.: 1, unused: 0 bucket size: 896 - 896 (first 896) 0(30173) hash = 128 fragments no.: 2, unused: 0 bucket size: 1024 - 1024 (first 1024) 0(30173) hash = 150 fragments no.: 1, unused: 0 bucket size: 1200 - 1200 (first 1200) 0(30173) hash = 193 fragments no.: 1, unused: 0 bucket size: 1544 - 1544 (first 1544) 0(30173) hash = 224 fragments no.: 1, unused: 0 bucket size: 1792 - 1792 (first 1792) 0(30173) hash = 258 fragments no.: 2, unused: 0 bucket size: 2064 - 2064 (first 2064) 0(30173) hash = 727 fragments no.: 1, unused: 0 bucket size: 5816 - 5816 (first 5816) 0(30173) hash = 831 fragments no.: 1, unused: 0 bucket size: 6648 - 6648 (first 6648) 0(30173) hash = 1024 fragments no.: 2, unused: 0 bucket size: 8192 - 8192 (first 8192) 0(30173) hash = 2049 fragments no.: 2, unused: 0 bucket size: 16384 - 32768 (first 20480) 0(30173) hash = 2056 fragments no.: 1, unused: 0 bucket size: 2097152 - 4194304 (first 2621440) 0(30173) hash = 2059 fragments no.: 1, unused: 0 bucket size: 16777216 - 33554432 (first 30805984) 0(30173) TOTAL: 181 free fragments = 33517712 free bytes 0(30173) ----------------------------- 0(30173) shm_mem_destroy 0(30173) destroying the shared memory lock 0(30173) terminating due to SIGCHLD -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHZ5doN4+o+2LtdFwRAknyAKCXJHhyHZlAooGoyONmFphFcZE3BgCfbJOO aN4wxnGPjLgxqVvx5YTgGac= =pkmq -----END PGP SIGNATURE-----

3 5

[Serusers] Newbie Problems: MySQL support in SER
by frq ser 19 Dec '07

19 Dec '07

Hi, Further to my previous email, there is another problem for which I need help. While trying to add MySQL support, I could not find the file src/sip_router/Makefile (as asked in the Getting started document (http://siprouter.onsip.org/doc/gettingstarted/ch11s05.html). The Makefile for the SER in the base directory (/lhome/.../ser/Makefile) does not refer to MySQL in the exclude_modules line (Note the ? after exclude_modules). Here is the excerpt from the Makefile: # if not set on the cmd. line or the env, exclude this modules: exclude_modules?= cpl ext extcmd \ postgres snmp \ im \ jabber mysql \ cpl-c \ auth_radius group_radius uri_radius avp_radius \ pa Does that mean that MySQL is already installed? How can I check if MySQL is already installed on my machine? I also found the following code in the Makefile but I am not sure what it means. .PHONY: dbinstall dbinstall: -@echo "Initializing ser database" scripts/ser_mysql.sh create -@echo "Done" . . . I'd be very grateful if anyone could help. Frq

1 1

[OpenSER-Users] Res: FIFO / SEMS
by Bruno Machado 19 Dec '07

19 Dec '07

Great job, Iancu. It is what I really needed. I tried to use the 'dialog' module and the session was torn down without any problems. Perfect. I have just one more question. I would like to log the generated BYE in ACC. Is it possible when OpenSER receive this BYE request from FIFO? Thanks a lot again bruno ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)lists.openser.org Enviadas: Segunda-feira, 17 de Dezembro de 2007 11:19:29 Assunto: Re: [OpenSER-Users] FIFO / SEMS Hi Bruno, In OpenSER 1.3 "BlackCat", the "dialog" module provides a way to terminate an ongoing dialog. See the generic docs: http://www.openser.org/docs/modules/1.3.x/dialog.html and about sending BYEs: http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN461 Regards, bogdan Bruno Machado wrote: > Hi all > > I am reading about FIFO and I am wandering in my thoughts, searching > for a solution > that tear down a session. I don't want to use Asterisk anymore for > this purpose. > Can I send a BYE message through FIFO? Should I use "t_uac_dlg"? I was > reading > about "t_uac_cancel" but it generates a CANCEL and, of course, it > won't resolve the problem. > > I didn't install SEMS yet. So, I would like to know if someone uses > this software > (OpenSER + SEMS) within this purpose. This link: > > http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.sw_prepaid_sip?op=file&rev… > <http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.sw_prepaid_sip?op=file&rev…> > > has this: > > "When the call duration exceeds the credit, the call is terminated by > SEMS." > > > So, I can realize it sends a BYE to OpenSER through FIFO and it will be > forwarded to the voip devices. Am I right? I will try to use the > 't_uac_dlg' now. > Any comments are welcome. > > Cheers > Bruno Machado > > > > ------------------------------------------------------------------------ > Abra sua conta no Yahoo! Mail > <http://br.rd.yahoo.com/mail/taglines/mail/*http://br.mail.yahoo.com/>, > o único sem limite de espaço para armazenamento! > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

1 0

[OpenSER-Users] Openser-1.3 packages are ready in Fedora testing repository
by Jan ONDREJ (SAL) 19 Dec '07

19 Dec '07

Hello, i am pleased to announce, that OpenSER has been approved as a part of Fedora distribution. openser-1.3.0 is ready in fedora-7-testing and fedora-8-testing repository. It will be moved to stable after aprox. a week. If you have openser < 1.3.0 installed on your fedora or EPEL, please turn off your automatic updates or update it manually before the final release. Follow these steps to update: read upgrading guide at: http://openser.org/dokuwiki/doku.php/install:1.2.2-to-1.3.0 run this command: yum --enablerepo=updates-testing update 'openser*' check your configuration files and just then restart openser Jan ONDREJ (SAL) PS: I am not a member of this mailinglist. If you wish to reply, please send email directly to me.

1 0

[OpenSER-Users] What is the difference?
by David Villasmil Govea 19 Dec '07

19 Dec '07

What's the difference between setbflag and setflag ?? Thanks David

2 2

[OpenSER-Users] Openser and Nokia E61
by David Villasmil Govea 19 Dec '07

19 Dec '07

Hello all, Anyone know how to connect a nokia e61 to openser? I've configured it to connect to asterisk and no problem whatsoever.... but seems impossible to connect to openser 1.2 Cheers David

3 2

[OpenSER-Users] need Help in TLS Problem in openser 1.3 Black CAT
by raviprakash sunkara 19 Dec '07

19 Dec '07

Hello Users Merry XMAS Resently I installed and Configure the openser 1.3 with TLS Support in FC6 with openSSL 0.9.8b Am Trying implementing the TLS of my OpenSER proxy First I trying TLS for Registering to My OpenSER Proxy Server. The Certificatation am using the default while in installing the openser 1.3 Am using eyebeam 1.5 with TLS. on the Screen am getting 503 service unavilable I don't know what happening in Server side. Please Help in this . -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead.

2 1

[OpenSER-Users] error_route not working
by Jayesh Nambiar 19 Dec '07

19 Dec '07

Hi All, I am using openser 1.3. Yesterday I added the error_route in my script to handle the parsing errors. But I see that the script logic is not going into error_route and I still see the message log filled up with messages like below: INFO:core:parse_first_line: bad message Dec 19 08:02:17 sw1 /usr/local/sbin/openser[26964]: ERROR:core:parse_msg: message=<INVITE sip:61: 394086416(a)207.251.84.106 SIP/2.0 If the error route is present in the script should'nt the script go to the error route in the case like above? My error_route is as follows: error_route { xlog("--- error route class=$(err.class) level=$(err.level) info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n"); xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n"); sl_send_reply("$err.rcode", "$err.rreason"); exit; } Can somebody give me some pointers on why error_route is not working in this case or am I missing something. Thanks in advance w/regards, Jayesh Save all your chat conversations. Find them online at http://in.messenger.yahoo.com/webmessengerpromo.php

1 0

[Serusers] nat problem
by Pezhman Lali 19 Dec '07

19 Dec '07

Dear I have a lot of problems with nat and registering, for example, grandstream does not work with a lot of internet providers, also linksys and ... sometimes the sip phone was registered, but the call is one way or no way. please help me, if my config has problem. Best Mani listen=1.1.1.1 debug=0 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL5 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="postgres://postgres:mypass@localhost:5432/test" alias="xxx.com" alias="xxx.com" alias="xxx.com" alias="xxx.com" alias="xxx.com" sip_warning=yes # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/postgres.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 1) modparam("acc", "db_url", "postgres://postgres:mypass@localhost:5432/test") modparam("acc", "db_missed_flag", 2) modparam("acc", "db_flag", 1) #modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("usrloc", "db_mode", 1) modparam("usrloc", "user_column", "user_id") modparam("usrloc", "db_url", "postgres://postgres:mypass@localhost:5432/test") # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("auth_db", "use_domain", 0) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "user_column", "user_id") modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "postgres://postgres:mypass@localhost:5432/test") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("rr", "enable_double_rr", 0) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 0) modparam("usrloc", "use_domain", 0) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock","unix:/var/run/rtpproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE" || method=="ACK") { setflag(1); }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method == "INVITE") { # if (!lookup("location")) { # xlog("L_ERR","%Tf: Rejecting %ru From %fu To %tu \n"); # sl_send_reply("404", "Not Registered"); # break; # }; if (!uri =~ "sip:43[0-9]{6}@*") { log(1, "Forwarding to Asterisk\n"); forward(2.2.2.2,5060); break; } } # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("xxx.com", "subscriber")) { www_challenge("xxx.com", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); #acc_db_request("404 missed call", "missed_calls"); break; }; setflag(2); }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

1 0

[Serusers] multi-domain support solution
by satish patel 19 Dec '07

19 Dec '07

Dear all I have solve my multidomain configuration with ser-0.9.6 and it is working fine for me and i am sending you my configuration for incomming users whos want do some stuff wid ser i have find many document on multi-support and i have consolidate all document and make my ser.cfg script or config cat /usr/local/etc/ser/ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.104.20 port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" #loadmodule "/usr/local/lib/ser/modules/call.so" #loadmodule "/usr/local/lib/ser/modules/vm.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("domain", "db_url", "mysql://root:mypasswd@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) #modparam("usrloc", "use_domain", 1) #modparam("registrar", "use_domain", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) #-----------Multi-domain Logic work blow in stanza--------# if (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; save("aliases"); break; }; #----Alias check--- # lookup("aliases"); # if (!uri==myself) { # route(1); # break; # }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #-----END---# Now to test configuration start ser #serctl start #serctl domain add abc.com #serctl add 222(a)abc.com tulip 222(a)abc.com #serctl add 333(a)abc.com tulip 333(a)abc.com this to user 222 and 333 for domain abc.com **Notes:- you should configure SRV recode on your DNS otherwise it will not work for you NOW configure your Xlite and try to call each other Then try to add new domain and do same thing which u have done before abc.com Thanks and Best of Luck $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Meet people who discuss and share your passions. Join them now.

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[Serusers] Fwd: Newbie Problems: ser.cfg
by frq ser 18 Dec '07

18 Dec '07

---------- Forwarded message ---------- From: frq ser <frq.ser(a)googlemail.com> Date: Dec 18, 2007 11:32 PM Subject: Newbie Problems: ser.cfg To: serusers(a)iptel.org Hi, I am trying to understand the process flow in the following ser.cfg file, which controls a SER proxy server at root@ip.address:5060. I have added "log()" command at almost every step to see how things proceed. The problem is that when I generate an OPTIONS message through SIPSAK (sipsak -T -vv -s sip:root@ip.address:5060), the process branches from route(0) to route(3), sends a message (200 Message received, probably from Sipsak), then comes back to route(0), completes this route, and then loops back to start of route(0) (as evident from the last log message "Inside main route..."). This is despite the "break" command in the main route block. Here's the output of the SER: [root@drone08 ~]# ser Listening on udp: 192.168.3.208 [192.168.3.208]:5060 tcp: 192.168.3.208 [192.168.3.208]:5060 Aliases: tcp: 192.168.3.208:5060 udp: 192.168.3.208:5060 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 5116 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(5114) Inside the main route.... 0(5114) Inside the main route.... 0(5114) just passed the sanity checks... 0(5114) inside method-register... 0(5114) inside method-options... 0(5114) Inside route3... 0(5114) End of route3... 0(5114) just past redir-route3.. 0(5114) Inside the main route.... The corresponding SIPSAK output is: C:\..Sipsak>sipsak -v -T -s sip:root@192.168.3.208:5060 warning: IP extract from warning activated to be more informational 0: 192.168.3.208 (0.000 ms) SIP/2.0 483 Too Many Hops 1: 192.168.3.208 (0.000 ms) SIP/2.0 201 Options message received, probably from Sipsak without Contact header On the other hand, when I send an INVITE message, this problem doesn't happen. The process completes the main route and doesn't loop back (ie I don't get the "Inside the main route..." message). [root@drone08 ~]# ser Listening on udp: 192.168.3.208 [192.168.3.208]:5060 tcp: 192.168.3.208 [192.168.3.208]:5060 Aliases: tcp: 192.168.3.208:5060 udp: 192.168.3.208:5060 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 4860 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(4858) Inside the main route.... 0(4858) just passed the sanity checks... 0(4858) inside method-register... 0(4858) outside methods-options... 0(4858) inside lookup-location... All I want to know is that when I send an OPTIONS msg, why the process doesn't terminate at the break command in the (if (method == "OPTIONS")) in the main route. Although it is not a critical problem, I just want to improve my understanding. I'll be glad if any of you could help. Fr. PS: The config file (ser.cfg) is given below: # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen= 192.168.3.208 #2.13 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # chg_rsf dec07 #loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/avpops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) #modparam("tm", "pass_provisional_replies", 1) route { log(1, "Inside the main route....\n"); # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; log(3, "just passed the sanity checks...\n"); # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); log(3, "inside method-register...\n"); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { log(3, "inside loose route...\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { log(3, "inside uri-myself...\n"); route(1); break; }; if (method=="ACK") { log(3, "inside method-ack...\n"); route(1); break; } else if (method=="REGISTER") { log(3, "inside method-register...\n"); route(2); break; }; if (method=="OPTIONS"){ log(3, "inside method-options...\n"); route(3); log(3, "just past redir-route3..\n"); break; log(3, "just past break...\n"); }; log(3, "outside methods-options...\n"); lookup("aliases"); if (uri!=myself) { log(3, "inside uri-myself 2....\n"); route(1); break; }; if (!lookup("location")) { log(3, "inside lookup-location...\n"); sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ log(3, "Inside route1...\n"); if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ log(3, "Inside route2...\n"); if (!save("location")) { #sl_reply_error(); sl_send_reply("201", "Register message received at Proxy Server"); }; } route[3]{ # ----------------------------------------------------------------------- # OPTIONS Message Handler # ----------------------------------------------------------------------- log(3, "Inside route3...\n"); if (!t_relay()){ sl_reply_error(); }; sl_send_reply("201", "Options message received, probably from Sipsak"); #forward(192.168.3.208, 5070); log(3, "End of route3...\n"); #t_release(); }

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[OpenSER-Users] Exposing the NIC IP's in openser.cfg
by Robert Dyck 18 Dec '07

18 Dec '07

I have been using openser as an ALG for the purpose of NAT traversal. Unfortunately my public IP is dynamically assigned. Although openser is aware of the IP's available on the host, I have searched unsuccessfully the documentation for a way to expose this information to the configuration script. Have I missed something? I am using the following ugly work around. DHCP client writes to /var/openser/my-ip From the rtpproxy init script - OPTIONS="-l 192.168.1.1/`cat /var/openser/my_ip` -s udp:localhost:12221" from openser.cfg - route{ # get our public ip, this happens only once if !$var(ip_initialized) == "yes" { xlog("L_NOTICE", "****************************************************\n"); xlog("L_NOTICE", "*** Initializing the IP for masquerading SIP Contact\n"); exec_avp("cat /var/openser/my_ip", "$avp(s:pub_ip)"); pv_printf("$var(my_ip)", "$avp(s:pub_ip)"); xlog("L_NOTICE", "*** New public IP is <$var(my_ip)>\n"); $var(ip_initialized) = "yes"; }; mangle contacts

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[OpenSER-Users] newbie help - handeling 302s and 404s
by Lazer Ramm 18 Dec '07

18 Dec '07

Hi, i need a simple config that achieves the following (w/ out a db) if a user is not registered then all calls should go to a default gw 192.168.10.1 the default gw will only respond w/ a 302 or a 404 if a 302 is received then openser should forward the call on to the new gw listed in the reply if a 404 is received the call should be forwarded to a different gw 172.16.20.1 if the send gw is done then the call should die. here is what i was thinking of but as a newbie i may be off my rocker i seem to get an error ERROR: t_check_status: cannot check status for a reply which has no T-state established but again my whole approach may be wrong. if (!lookup("location")) { rewritehostport ("192.168.10.1:5060"); if (t_check_status("(404)|(480)")) { log("404 or 480 negative reply\n"); rewritehostport ("172.16.20.1:5060"); } else{ t_on_failure("1"); t_relay(); } failure_route[1] { get_redirects("*"); t_relay(); } } thanx in advance

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Re: [OpenSER-Users] Want to know about configuring openser as strict router
by Vikas CG. 18 Dec '07

18 Dec '07

Hi, Thanks for the information, Actually we were looking to some server which can acts as a strict router. Can u suggest regarding this? Thanks, Vikas C.G. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Tuesday, December 18, 2007 7:59 PM To: Vikas CG. Cc: users(a)openser.org Subject: Re: [OpenSER-Users] Want to know about configuring openser as strict router Hi! Openser is a loose-router. This means that the inserted Record-Route header do have the ;lr parameter. Thus, other SIP clients know that openser is a looserouter and weill send messages in loose-router syntax. On the other hand, openser checks if the next route header and/or the RURI contains the ;lr parameter to make conversion from/to strict-router syntax. Thus, depending on if the previous and next hops are strict or loose routers, openser will do the conversion (by using the loose_route() function). regards klaus Vikas CG. schrieb: > Hi all, > > > > We are using openser server for testing. Can we configure it as > strict router? > > I have tried making changes in configuration file by enabling Record > Route. > I want to know whether the openser will strictly acts as "Strict Router" > I.e. performing actions like modifying Request URI and last route header. > Can any body suggest me? > > > > Thanks in advance , > Vikas C.G. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] OpenSER: AVP script flag bugs in 1.3 and trunk
by Helmut Kuper 18 Dec '07

18 Dec '07

Hello, in openser 1.3 and current trunk the AVP script flags still not work as expected. So here again the fixes I did to solve that problems in 1.3 and trunk: ... Hi again, further tests with avp and script_flags raised new problems. When I write resp. set an AVP like "$avp(i1:10)=10; $avp(i2:10)=20;" and checking them with "xlog('L_INFO', 'avp(i1:10) avp(i1:20)\n');" it leads to "20 20" as output. In file pvar.c in function pv_get_avp_name() I had to replace *name_type = 0; with *name_type = ip->pvn.u.isname.type; Then that worked. - -.- Additionally I fixed the Alias-Duplicate Problem when you do something like avp_aliases("A=i1:10; B=i2:20"); In file usr_avp.c I had to change line 427 in function check_avp_galias() from type &= AVP_NAME_STR; to type &= (AVP_NAME_STR|AVP_SCRIPT_MASK); - -.- Then at last, if u want assign a value to an AVP-Alias which uses script flags replace in file usr_avp.c in function add_avp_galias() around line 482 the line ga->avp.type = type&AVP_NAME_STR; with ga->avp.type = type&(AVP_NAME_STR|AVP_SCRIPT_MASK); regards Helmut

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[OpenSER-Users] Want to know about configuring openser as strict router
by Vikas CG. 18 Dec '07

18 Dec '07

Hi all, We are using openser server for testing. Can we configure it as strict router? I have tried making changes in configuration file by enabling Record Route. I want to know whether the openser will strictly acts as "Strict Router" I.e. performing actions like modifying Request URI and last route header. Can any body suggest me? Thanks in advance , Vikas C.G.

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[OpenSER-Users] Question about feature requests
by Sergio Gutierrez 18 Dec '07

18 Dec '07

Hello everybody. I would like to know how is the right procedure to implement a new feature for OpenSER. For example, if I propose a new Feature, is it well to include the patch for the implementation, or I just propose it, and wait for the approval of one of the core developers? Best regards. Sergio Gutiérrez.

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[OpenSER-Users] ERROR:core:io_watch_add: epoll_ctl failed: Invalid argument [22]
by Mihail Gluhowchenko 18 Dec '07

18 Dec '07

Truing to start openser-1.3.0. In log i`m saw: ERROR:core:io_watch_add: epoll_ctl failed: Invalid argument [22] Dec 18 12:37:17 devel s-eltex[27564]: CRITICAL:core:tcp_main_loop: failed to add process 5 (TCP main) unix socket to the fd list Dec 18 12:37:17 devel s-eltex[27564]: CRITICAL:core:tcp_main_loop: exiting... In config i`m set listen=udp:192.168.138.59:5060 Why openser truing to open tcp sockets? And another bug: If stop openser and then start. If dont switch off SIP clients server down with error.

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[Serusers] ser multi-domain support SOLVED
by satish patel 18 Dec '07

18 Dec '07

Dear all I have solve my multidomain configuration with ser-0.9.6 and it is working fine for me and i am sending you my configuration for incomming users whos want do some stuff wid ser i have find many document on multi-support and i have consolidate all document and make my ser.cfg script or config cat /usr/local/etc/ser/ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.104.20 port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" #loadmodule "/usr/local/lib/ser/modules/call.so" #loadmodule "/usr/local/lib/ser/modules/vm.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("domain", "db_url", "mysql://root:mypasswd@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) #modparam("usrloc", "use_domain", 1) #modparam("registrar", "use_domain", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) #-----------Multi-domain Logic work blow in stanza--------# if (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; save("aliases"); break; }; #----Alias check--- # lookup("aliases"); # if (!uri==myself) { # route(1); # break; # }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #-----END---# Now to test configuration start ser #serctl start #serctl domain add abc.com #serctl add 222(a)abc.com tulip 222(a)abc.com #serctl add 333(a)abc.com tulip 333(a)abc.com this to user 222 and 333 for domain abc.com **Notes:- you should configure SRV recode on your DNS otherwise it will not work for you NOW configure your Xlite and try to call each other Then try to add new domain and do same thing which u have done before abc.com Thanks and Best of Luck $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Share files, take polls, and discuss your passions - all under one roof. Click here.

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[Serusers] call.so and vm module missing
by satish patel 18 Dec '07

18 Dec '07

Dear all I have compileing ser-0.9.6 and it is compiled successful but when i going to /usr/local/lib/ser/module directory there is no required module call.so and vm.so how to compile i have also try this one make modules=modules/call modules but there is no file found How to compile and get support of call.so module and vm.so module $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

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[Serusers] Ser sends messages not in that order in which it has received
by Alexandr Volkov 18 Dec '07

18 Dec '07

Hi, We have found one unpleasant situation with a Ser often shown at greater loadings. At greater loadings and at work of a Ser with number of child processes greater than one (Parameter 'children' in ser.cfg), Ser sends messages not in that order in which it has received. And as the result appears request pending errors. We simulated this situation continuously reinviting the remote UA. Usually the error is occured on the tenth-twentieth reinvite. In the attachment there are SIP logs reproducing this situation. Somebody can tell in occasion of this problem, or offer the decision? Or whether there is a version in which there is no this lack? Alexandr

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[OpenSER-Users] OpenSER - unixODBC Module
by Martin.Gruber1＠infineon.com 18 Dec '07

18 Dec '07

Hi, Well I now know my problem why I cannot connect to my Oracle-Database. The Connectionport hast changed. My question is, how can I insert the Port-Number the the connectionstring in openser.cfg....... unixodbc://user:password@oracle.server.com:PortNr/ORA Does not work at all. Does anybody know, where I can add the Port Number in the Connectionstring? Kind regards, Martin

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Re: [OpenSER-Users] OpenXcap exception: TypeError in twisted.web2.stream.IByteStream
by Antoine Fressancourt 18 Dec '07

18 Dec '07

Hello, I have the same error as Gregorio on my installation, I am running OpenSER 1.3 and my XCAP client is eyeBeam². Thanks in advance for your help, Antoine 2007/10/30, Gregorio <goyo.villalba(a)dialcom.com>: > > Hi all. > > > > I get an exception in openXcap when I send a GET to retrieve the resources > file. > > I explain the process I followed: > > > > ----->Send GET to openXcap. > > <----Receive 404, because there is not file. > > > > -----> Send PUT to openXcap. The body contains a resource-list.xmldocument 'with no data': > > Body; > > <?xml version='1.0' encoding='UTF-8'?> > > <resource-lists xmlns='urn:ietf:params:xml:ns:resource-lists' > xmlns:cp='urn:ietf:params:xml:ns:copycontrol'> > > <list></list> > > </resource-lists> > > > > <----Receive 200 OK. So openXcap likes my document. > > > > -----> Send GET to openXcap. > > <---- Receive 500 "An error occurred rendering the requested page. More > information is available in the server log". So the same document that was > accepted in the PUT throws an exception. > > > > The server log says: > > > > Oct 30 15:01:55 localhost openxcap[2791]: [-] Exception rendering: > > Oct 30 15:01:55 localhost openxcap[2791]: [-] Traceback (most recent call > last): > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/posixbase.py", line 228, > in mainLoop > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self.runUntilCurrent > () > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/base.py", line 533, in > runUntilCurrent > > Oct 30 15:01:55 localhost openxcap[2791]: [-] f(*a, **kw) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 239, in > callback > > Oct 30 15:01:55 localhost openxcap[2791]: [-] > self._startRunCallbacks(result) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 304, in > _startRunCallbacks > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self._runCallbacks() > > Oct 30 15:01:55 localhost openxcap[2791]: [-] --- <exception caught > here> --- > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 317, in > _runCallbacks > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self.result = > callback(self.result, *args, **kw) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/xcap/resource.py", line 51, in > sendResponse > > Oct 30 15:01:55 localhost openxcap[2791]: [-] response = > http.Response(response.code, stream=response.data) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/web2/http.py", line 109, in > __init__ > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self.stream = > IByteStream(stream) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] exceptions.TypeError: > ('Could not adapt', array('c', "<?xml version='1.0' > encoding='UTF-8'?><resource-lists > xmlns='urn:ietf:params:xml:ns:resource-lists' > xmlns:cp='urn:ietf:params:xml:ns:copycontrol'><list></list></resource-lists>"), > <InterfaceClass twisted.web2.stream.IByteStream>) > > > > > > Anyone know what is wrong? > > Thanks and best regards. > > Gregorio > > > > > > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

2 1

[OpenSER-Users] publish with empty body
by jh＠tutpro.com 18 Dec '07

18 Dec '07

while testing a new sip ua, i noticed that at some point it sent (don't ask me why because i don't know) a publish request without any body. it was not an initial publication, because it contained SIP-If-Match header with an etag. the publication was for even package presence. what openser presence server did was that it accepted the publish with 200 ok that contained a NEW etag value in its SIP-If-Match header. i have been trying to find out from rfcs where this kind of behavior would have been defined, but so far without success. rfc3903 says: 5. The ESC processes the published event state contained in the body of the PUBLISH request. If the content type of the request does not match the event package, or is not understood by the ESC, the ESC MUST reject the request with an appropriate response, such as 415 (Unsupported Media Type), and skip the remainder of the steps. in this publish content type didn't match the event package, because there was none. reading further i found this: * If the request has no message body and contained no entity-tag, the ESC SHOULD reject the request with an appropriate response, such as 400 (Invalid Request), and skip the remainder of the steps. Alternatively, in case either ESC local policy or the event package has defined semantics for an initial publication containing no message body, the ESC MAY accept it. but it does not apply, because the publication was not initial. so, have i found a bug or any other ideas? -- juha

2 2

Re: [Serusers] rewritehostport and RFC3261 compliant
by Klaus Darilion 18 Dec '07

18 Dec '07

Hi Adrien! I Cc this email also to the seruser mailing list as it might be interesting for others too. Adrien Lemoine schrieb: > Hi Klaus, > > I've some good news !!! (Sorry to take your time) > > I try the solution in the URL paste in the last mail, and it runs ! > > So I've something like this : > > if ( t_check_status("(1[0-7][0-9])|(180)") ) { > xlog("L_WARN","time=[%Tf] call id=[%ci] call seq=[%cs] contact > header=[%ct] from uri=[%fu] from tag=[%ft] request's method=[%rm] request's > uri=[%ru] to uri=[%tu] to tag=[%tt] sip message id=[%mi] process id=[%pp] ip > source=[%is] flags=[%mf], fix To-Tag failure [Thomson]\n"); > subst('/^(To:.*)\;tag=[^;[:cntrl:]]*/\1/i'); > } > break; > > The guy precise that 183 don't have to be included in this test, why ? Bogdan was afraid that this could break "early media", but I'm sure it is best to include 183 as well. Thus this regular expression should handle all provisional responses. t_check_status("(1[0-9][0-9])") > Is this could be okay for simplify the solution and don't take the risk to > break an other service by collateral effects ? You never know and it would be better to fix it in the SIP phone/gateway. But I think it is a acceptable workaround. > I think I beging to understand the problem : my Thomson product "remember" > of the first to-tag include in "180 ringing" and doesn't accept an other > to-tag for the same transaction. Isn't it ? Yes. Some theory: SIP has to match incoming requests and responses to existing dialog. This is done by comparing the 3-tuple call-id/from-tag/to-tag. Messages which are sent out of dialog do not have a to-tag (tag parameter in the To: header). The initial INVITE request is sent "out-of-dialog" (no to-tag). If a SIP client responses to the INVITE request it adds a to-tag to the response. Thus, if the INVITE is sent to multiple devices (e.g. parallel ringing on multiple SIP devices or call forwarding on timeout) the caller will receive multiple responses. As the to-tag is different for each called SIP device the caller can detect responses from different devices. (this is a feature of SIP). If the caller SIP client receives provisional responses (101-199) it will create an "early dialog". Once a final response is received the dialog will convert into "confirmed" state ( 200 OK) or the dialog will be terminated (3xx-6xx responses). Example: caller proxy phone1 phone2 *0 --INVITE----------> <-100 (no to tag)-- ----INVITE------> *1 <---100 trying--- <---180 ringing-- <---180 ringing----- *2 ----INVITE-----------------> <---100 trying-------------- <---180 ringing------------- <---180 ringing----- *3 <---200 ok----------------- <----200 ok --------- *4 ----CANCEL------> ........ An RFC compliant SIP client should handle the scenario following: *0: the caller sends the INVITE (out-of-dilaog). *1: The caller receives a 100 trying response. A "100 trying" does not have a to-tag and thus does not create dialogs. It will only stop retransmissions. *2: The caller receives a provisional response. It creates an "early-dialog" with the identifier call-id/fromtag/totag1. *3: The caller receives another provisional response. The response has a different to-tag. Thus the caller creates a second "early-dialog" with identifier call-id/fromtag/totag2. *4: The caller receives a final response. The 200 ok matches call-id/fromtag/totag2. Thus, the first early-dialog will be terminated and the second early-dialog will be transformed into "confirmed" state. Some non-complient clients handle it in other ways, e.g.: - Asterisk with "pedantic=no": Asterisk does not care about totags in provisional responses but only in the 200 ok. (works but insecure). - Asterisk with "pedantic=yes": Asterisk learns the totag from the first provisional response. All further responses with different totag are ignored. (breaks compatibility). I guess the Thomson device has the some bug as Asterisk in pedantic mode. > If it's that, I don't read something about this behaviour in the RFC to > argument my quick fix. Of course the workaround with "removing totags in provisional responses" breaks the RFC - but in the case with broken clients the RFC is already broken by the clients. > An other problem appears, you said that "try to remove the to-tag in the 180 > and 183 responses *to this client*". I would like too, but how can I match > UserAgent in Status, there's not. If the INVITE has a user agent header like "User-Agent: Thomsen abcd....." you activate the reply_route. E.g. ... if ((method == INVITE) && (search("User-Agent: Thomson")) { t_on_reply("1"); xlog("L_INFO","buggy Thomson, activate reply route to strip totag"); } ... onreply_route[1] { xlog("L_INFO","reply route: strip totag"); # put some more xlog here if wanted... if (t_check_status("(1[0-9][0-9])")) { subst('/^(To:.*)\;tag=[^;[:cntrl:]]*/\1/i'); } } regards Klaus

1 0

[OpenSER-Users] Using append_hf("Route: xx") and loose_route() not stripping pre-loaded routeset?
by Taisto Qvist 18 Dec '07

18 Dec '07

Hi Guys, I have been trying all day, to create "static routing" in my openser-proxy(1.3). My openser-script is based on the default, so what I am doing is just: ------------------------------------------- if ( loose_route() ) { // strip of preloaded route is NOT being done here?? route(); } if ( isToMe() ) { // Do local logic } else { // forward this via proxyX append_hf("Route: <sip:1.1.1.1:5060;lr>\r\n"); if (loose_route() ) { route(1); } else { // I always end up here...loose_route() always returns false... // because its looking at the first route, that should have been stripped off... } ----------------------------- I have added lots of traces to lookup.c, and according to those, it seems that the preloaded route is not stripped off properly? Once I get too the second "loose_route()", that function IS STILL looking at the first preloaded route, which should have been stripped in the first "if". I am getting the feeling that the "deletion/stripping" of the first, preloaded, Route, is not done directly, causing the second check, to find the first... My work-around is naturally to use setdsturi(), but I thought that loose_route() should be able to do all this for me...? Also, this used to work, in version 1.1, about year ago :-) Any toughts? Regards Taisto Qvist

2 3

[Serusers] SIP Express Bundle - let's get started
by Greger V. Teigre 18 Dec '07

18 Dec '07

Hi guys, thanks a bunch for lots of input and I really appreciate the willingness to contribute. Thus, I have created a "project page": http://www.iptel.org/sip_express_bundle_sip_service_in_15_minutes I thought we could gather the current perspective on that page and document our decisions as we go. It will hopefully be useful in our process, as well as documentation for us and others later. I have noted the following volunteers: Jai: testing and installation work ram: testing SIP: tesing Jiri: anyting? Mike: testing and documentation Maybe we should set up a small mailing list for coordination emails, but for now, let's use serusers (where I think all the comments came). (I copy the other lists on this post, so the other lists know that the discussion will move to serusers). Out of the comments, I read CentOS and vmware as the most wanted combination. I have documented the pros and cons on the project page, and suggest that we do some testing before we decide. I also tried out rpath (which I have no previous experience with). My observations are documented on the same page. Thus, I have started setting up a minimal CentOS virtual appliance found in vmware's appliance directory on an esx server. I will send details on accessing it to the volunteers once it is up and running (decompressing, unpacking, and building a non-split disk takes an awful lot of time :-( ). Ok, further comments, ideas, etc, please post to serusers or edit/add comments to the project page (requires an iptel.org account). I see the following steps with documentation as we go (steps also found on the project page): 1. Testing and specification of what we want to accomplish 2. Environment and OS setup to ensure that we easily can release new versions 3. Installation and configuration of the software. I assume this step also will involve development of some tools we need, as well as adaptation of existing stuff 4. Testing and user documentation 5. Packaging and deployment Let's get going! g-)

12 44

[OpenSER-Users] invite resend bug?
by yoshiki 18 Dec '07

18 Dec '07

hi, I am usingo openser-1.2.0 When a client transmitted INVITE and transmitted INVITE once again because a reply of openser was slow, these two INVITE message was treated as another message. The peer transmitted 200OK for INVITE of the beginning, but a server transmits "404 not found" to next INVITE. Will even a message of the same call ID be handled in this way? yokkie -------------------------------------- New Design Yahoo! JAPAN 2008/01/01 http://pr.mail.yahoo.co.jp/newdesign/

1 0

[OpenSER-Users] Preparation for 1.2.3 release
by Bogdan-Andrei Iancu 17 Dec '07

17 Dec '07

Hi, I would like to try to get 1.2.3 out before Christmas, so here is a starting plan: 1) do some backporting related to bug fixes (like in TM, with Route for local generated requests) 2) do some backporting related to compiling errors and warnings (solaris, 64 bits, BSD, etc) 3) backport the new default configuration script we have in 1.3.0 I can take care of all this, not a problem. Is anybody seeing any stopper for doing the release by Thursday 20? Regards, Bogdan

1 0

[OpenSER-Users] OpenSER 1.3.0 release - updates
by Bogdan-Andrei Iancu 17 Dec '07

17 Dec '07

Hi, The sources are no available for download from the SF site also. See: https://sourceforge.net/project/showfiles.php?group_id=139143 In some case SF will be better to use (for this purpose) as they mirror the files in multiple locations. Also, I generated and uploaded on openser.org packages for debian unstable and testing. See: http://www.openser.org/pub/openser/latest/packages/debian-unstable/ During next week, packages for debian stable will be generated and uploaded on the project local apt repository. Also some other binary tarballs are promised (by other contributors) for different OS and archs. Regards, Bogdan

1 1

[OpenSER-Users] FIFO / SEMS
by Bruno Machado 17 Dec '07

17 Dec '07

Hi all I am reading about FIFO and I am wandering in my thoughts, searching for a solution that tear down a session. I don't want to use Asterisk anymore for this purpose. Can I send a BYE message through FIFO? Should I use "t_uac_dlg"? I was reading about "t_uac_cancel" but it generates a CANCEL and, of course, it won't resolve the problem. I didn't install SEMS yet. So, I would like to know if someone uses this software (OpenSER + SEMS) within this purpose. This link: http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.sw_prepaid_sip?op=file&rev… has this: "When the call duration exceeds the credit, the call is terminated by SEMS." So, I can realize it sends a BYE to OpenSER through FIFO and it will be forwarded to the voip devices. Am I right? I will try to use the 't_uac_dlg' now. Any comments are welcome. Cheers Bruno Machado Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

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Re: [OpenSER-Users] OpenXcap fails to start the HTTP on port
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

Hi all , I have a problem in running openxcap server this is my syslog Starting Open XCAP 0.9.7 Dec 17 21:02:03 ankith openxcap[31833]: [-] Traceback (most recent call last): Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/bin/openxcap", line 56, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from xcap.server import XCAPServer Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/xcap/server.py", line 16, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2 import channel, resource, http, responsecode, server Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/channel/__init__.py", line 7, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2.channel.cgi import startCGI Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/channel/cgi.py", line 8, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2 import http, http_headers, server, responsecode Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/http.py", line 27, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2 import iweb Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/iweb.py", line 58, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] class IOldNevowResource (components.Interface): Dec 17 21:02:03 ankith openxcap[31833]: [-] AttributeError: 'module' object has no attribute 'Interface' I think problem in somewhere in zope interface ..............but I have install zope 3.3.0 version ......... please help me out ............ Regards Amit On Wed, 2007-10-17 at 13:11 +0200, Gregorio wrote: > Hi all. > > > > Finally (buff!!) I could configure openXcap, after executing ‘openxcap > start’ the log says: > > > > ‘ [-] Log opened. > > [-] Starting Open XCAP 0.9.4 > > [-] Supported Root URIs: http://192.168.1.206 > > [-] twisted.web2.channel.http.HTTPFactory starting on 8000’ > > > > And nothing else, no more logs displayed. OpenXcap process is running > in the machine: > > [root@XXXXX]# ps -ef|grep xcap > > root 926 1 0 18:02 ? > 00:00:00 /usr/bin/python /etc/rc.d/init.d/openxcap start > > > > But no one is listening in port 8000, neither before executing xcap > nor after. > > > > I haven´t found any error in any log so I don’t know what is going on. > > Anyone can help me? > > > > Thanks. > > Gregorio > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

1 0

Re: [Serusers] SER 2.0 installation - SER does not start due to missing modules
by Strassburger, Patrick 17 Dec '07

17 Dec '07

Hello Greger, I missed the mysql-devel package. Nevertheless now it claims on folowing: : [root@ser ser]# ser -c 0(9562) ERROR: load_module: could not open module </usr/local/lib/ser/modules/x mlrpc.so>: /usr/local/lib/ser/modules/xmlrpc.so: cannot open shared object file: No such file or directory 0(9562) parse error (75,13-14): failed to load module 0(9562) set_mod_param_regex: No module matching <xmlrpc> found 0(9562) parse error (142,20-21): Can't set module parameter 0(9562) parse error (235,16-17): unknown command, missing loadmodule? I installed php-xmlrpc-5.1.6-5.el5.i386.rpm and did another installation but still the same. (I did the following command for modules: "make group_include="standard mysql" all", "make group_include="standard mysql" installall") Do I miss anything here? Kind regards Patrick -----Ursprüngliche Nachricht----- Von: Greger V. Teigre [mailto:greger@teigre.com] Gesendet: Montag, 17. Dezember 2007 15:12 An: Strassburger, Patrick Cc: serusers(a)lists.iptel.org Betreff: Re: [Serusers] SER 2.0 installation - SER does not start due to missing modules You must do make install with the same group_include. Verify that you have mysql-devel and mysql rpm packages installed (required to compile and run). g-) Strassburger, Patrick wrote: Hello all, I am a newbie in SER and am failing since a few days installing SER 2.0. After several installation attempts I would like to ask you for some help. I did a "make group_include="standard mysql" all" but cannot see module mysql - where do I find those two modules and can I add them afterwards? Kind regards Patrick I get the following error and do not know how to resolve this: [root@ser ser]# ser -c 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: cannot open shared object file: No such file or directory 0(5613) parse error (55,13-14): failed to load module 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/xmlrpc.so>: /usr/local/lib/ser/modules/xmlrpc.so: cannot open shared object file: No such file or directory 0(5613) parse error (75,13-14): failed to load module 0(5613) set_mod_param_regex: No module matching <xmlrpc> found 0(5613) parse error (142,20-21): Can't set module parameter 0(5613) parse error (235,16-17): unknown command, missing loadmodule? ERROR: bad config file (4 errors) [root@ser ser]# ls -l /usr/local/lib/ser/modules/ total 4944 -rwxr-xr-x 1 root root 67012 Dec 17 15:02 acc_db.so -rwxr-xr-x 1 root root 73256 Dec 17 15:02 acc_syslog.so -rwxr-xr-x 1 root root 69954 Dec 17 15:02 auth_db.so -rwxr-xr-x 1 root root 58083 Dec 17 15:02 auth.so -rwxr-xr-x 1 root root 37353 Dec 17 15:02 avp_db.so -rwxr-xr-x 1 root root 168776 Dec 17 15:02 avpops.so -rwxr-xr-x 1 root root 79122 Dec 17 15:02 avp.so -rwxr-xr-x 1 root root 279393 Dec 17 15:02 ctl.so -rwxr-xr-x 1 root root 72908 Dec 17 15:02 db_ops.so -rwxr-xr-x 1 root root 63015 Dec 17 15:02 dispatcher.so -rwxr-xr-x 1 root root 21612 Dec 17 15:02 diversion.so -rwxr-xr-x 1 root root 114544 Dec 17 15:02 domain.so -rwxr-xr-x 1 root root 45712 Dec 17 15:02 enum.so -rwxr-xr-x 1 root root 84369 Dec 17 15:02 exec.so -rwxr-xr-x 1 root root 81406 Dec 17 15:02 fifo.so -rwxr-xr-x 1 root root 53450 Dec 17 15:02 flatstore.so -rwxr-xr-x 1 root root 43426 Dec 17 15:02 gflags.so -rwxr-xr-x 1 root root 87582 Dec 17 15:02 lcr.so -rwxr-xr-x 1 root root 35891 Dec 17 15:02 maxfwd.so -rwxr-xr-x 1 root root 112643 Dec 17 15:02 mediaproxy.so -rwxr-xr-x 1 root root 87493 Dec 17 15:02 msilo.so -rwxr-xr-x 1 root root 187830 Dec 17 15:02 nathelper.so -rwxr-xr-x 1 root root 21211 Dec 17 15:02 options.so -rwxr-xr-x 1 root root 131901 Dec 17 15:02 pdt.so -rwxr-xr-x 1 root root 272121 Dec 17 15:02 permissions.so -rwxr-xr-x 1 root root 122166 Dec 17 15:02 pike.so -rwxr-xr-x 1 root root 26180 Dec 17 15:02 print.so -rwxr-xr-x 1 root root 47605 Dec 17 15:02 ratelimit.so -rwxr-xr-x 1 root root 152851 Dec 17 15:02 registrar.so -rwxr-xr-x 1 root root 114904 Dec 17 15:02 rr.so -rwxr-xr-x 1 root root 63288 Dec 17 15:02 sanity.so -rwxr-xr-x 1 root root 66950 Dec 17 15:02 sl.so -rwxr-xr-x 1 root root 34423 Dec 17 15:02 speeddial.so -rwxr-xr-x 1 root root 120815 Dec 17 15:02 textops.so -rwxr-xr-x 1 root root 43284 Dec 17 15:02 timer.so -rwxr-xr-x 1 root root 848970 Dec 17 15:02 tm.so -rwxr-xr-x 1 root root 117605 Dec 17 15:02 uac.so -rwxr-xr-x 1 root root 38801 Dec 17 15:02 unixsock.so -rwxr-xr-x 1 root root 38987 Dec 17 15:02 uri_db.so -rwxr-xr-x 1 root root 38611 Dec 17 15:02 uri.so -rwxr-xr-x 1 root root 351480 Dec 17 15:02 usrloc.so -rwxr-xr-x 1 root root 100362 Dec 17 15:02 xlog.so [root@ser ser]# ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900. ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 17 Dec '07

17 Dec '07

At 09:10 AM 12/17/2007, Greger V. Teigre wrote: >Sounds good, Mike. >As we use berlios.de, there's really no official CVS maintainer, so >my comment is as good as any. That is a stable CVS. I only asked for maintainer as standard courtesy to be able to check-in ongoing documentation for the project if such materials are to be stored there. >Why do you need a mirror? Only offered if team see a need for a separate site for safety or separate "read-only" distribution overhead. >I'm sure we can create a new cvs module in the official repository >(though we may want to evaluate whether to use berlios' subversion instead). My best developers friends in US and CA are in various stages of moving from old CVS to subversion. It started as personal preference based on ease-of-use. Functionality of SVN is moving onward and correcting some traditional 'pain' associated with CVS when used by multiple active users. Client side tools available and growing too. >If we use subversion, you can use svk to create a mirror. Yup. "It supports repository mirroring, disconnected operation, history-sensitive merging, and integrates with other version control systems, as well as popular visual merge tools. " A PERL script to automate smart mirroring within your own subversion site was made available in 2005. I do not know how widely it is used. But my guys liked it as a way to be even more selective. http://subversion.tigris.org/servlets/ReadMsg?listName=users&msgNo=32596&at… >g-) > >Mike Trest - Personal wrote: >>Greger, et. al. >> >>While I did volunteer for Documentation and Testing already, >>I am going to branch out and perform each step myself >>so we can "document as we design". >> >>To that end I am setting up two "target machines" in New York on which >>to install-test-document each step wearing the hat of the Enterprise-IT-Guy. >>I have a couple of techs that we can use in the role of "outside tester" who >>will strictly follow the documentation to give the procedures a >>good Q-A test. >>I can be ready in early Janurary. >> >>I would like to hear from the current CVS maintainer regards creation >>of third machine as a Subversion / CVS mirror. Can you contact me >>via EMAIL with some guidance? >> >>Regards, ..mike.. >> >>At 03:03 AM 12/16/2007, Greger V. Teigre wrote: >>>Using kickstart sounds good to me. I think we should keep >>>separate the different steps: >>>1. Create a base image of the OS >>>2. Install the iptel.org apps and dependencies >>>3. Do local configurations >>>4. Create deployable virtual machine >>> >>>If each of these steps are isolated, different people can be >>>responsible for the different steps. >>> >>>I have reorganized the project page into sections and incorporated >>>lots of input from Jeremy at >>><http://iptel.org/bundle/project>http://iptel.org/bundle/project >>>We now need to create a project plan, split into responsibilities, >>>and take some key decisions. >>>I think one key decision that we have already converged on is: for >>>first version use CentOS5 and create vmware appliance. >>>g-) >>> >>> >>>Jai Rangi wrote: >>>>Sounds good, >>>>Here is the script. Its been a while that we have used it. We >>>>might need to modify something. I can look into it later. Let me >>>>know if this is something you are looking for. >>>>************************ >>>>#!/bin/sh >>>># Need to add /bin/sh path >>>># This script will install the softwares required to build softswitch. >>>># Sysrequirements FC5, DELL 1950, >>>># Jai >>>># 12/11/06 >>>> >>>>HOSTNAME=`hostname` >>>>TEMPLATE=<http://192.168.2.201/> 192.168.1.1 >>>> >>>>if [ "`id -u`" != "0" -o `hostname` != "$HOSTNAME" ]; then >>>> printf "NEED TO BE ROOT AND TO RUN ON $HOSTNAME\n" >>>> exit 0 >>>>fi >>>> >>>>echo "You are ROOT on Make sure you know what are you doing, >>>> Press ENTER to continue" >>>> >>>>read >>>> >>>>echo "Server name = $HOSTNAME" >>>>echo "FTP Server = $TEMPLATE" >>>>echo "Press ENTER to start" >>>> >>>>read >>>> >>>>printf "PROCEEDING WITH INSTALLATION OF Softswitch on ${HOSTNAME} ...\n\n" >>>> >>>> >>>>printf "ARE YOU SURE YOU WANT TO CONTINUE (Y/N)?" >>>>read X >>>>if [ "x$X" != "xY" ]; then >>>> printf "aborting. Press Y or N \n" >>>> exit 0 >>>>fi >>>>printf "\n" >>>> >>>>mkdir //home/switch >>>>ln -s /home/switch /usr/local/ >>>> >>>> >>>>cd /usr/local/src >>>> >>>>echo "------INSTALLING freeradius -----------------------------" >>>> >>>>printf "Do you want to INSTALL FREERADIUS (Y/N)?" >>>>read X >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><ftp://ftp.freeradius.org/pub/radius/old/freeradius-1.1.2.tar.gz>ftp://ftp.freeradius.org/pub/radius/old/freeradius-1.1.2.tar.gz >>>> >>>>tar zxvf freeradius-1.1.2.tar.gz >>>>cd freeradius-1.1.2 >>>>./configure >>>> >>>>make >>>>make install >>>>#cd .. >>>>echo "enable digest auth in radiusd.conf, configure >>>>clients.conf, create radius db and import tables." >>>> >>>>else >>>>printf "OK YOU Pressed N that means you don't want free radius. >>>>No Problem \n" >>>>fi >>>> >>>>echo "------ radius client lib -----------------------------" >>>>printf "Do you want to radius client lib (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><http://download.berlios.de/radiusclient-ng/radiusclient-ng-0.5.2.tar.gz>http://download.berlios.de/radiusclient-ng/radiusclient-ng-0.5.2.tar.gz >>>> >>>> >>>>tar zxvf radiusclient-ng-0.5.2.tar.gz >>>>cd radiusclient-ng-0.5.2 >>>>./configure >>>>make >>>>make install >>>> >>>>#vi /etc/ld.so.conf >>>># /usr/local/lib >>>>cat "/usr/local/lib" > /etc/ld.so.conf.d/radius- i386.conf >>>> >>>>ldconfig >>>>cd .. >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you don't want radius >>>>client lib. I don't care, I am just a script \n" >>>> >>>>fi >>>> >>>>echo "------ MEDIAPROXY -------------------" >>>>printf "Do you want to Mediaproxy (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><http://mediaproxy.ag-projects.com/old/mediaproxy-1.4.2.tar.gz>http://mediaproxy.ag-projects.com/old/mediaproxy-1.4.2.tar.gz >>>>tar mediaproxy-1.4.2.tar.gz >>>>mv mediaproxy /usr/local/mediaproxy >>>> >>>>#cp mediaproxy.ini.sample mediaproxy.ini >>>> >>>>#vi mediaproxy.ini >>>># edit the options as per our envi. >>>>cd /usr/local/mediaproxy/ >>>>wget <ftp://$TEMPLATE/configure_ser>ftp://$TEMPLATE/configure_ser >>>>_server/configfiles/mediaproxy.ini >>>> >>>>cp /usr/local/mediaproxy/boot/mediaproxy.other /etc/init.d/mediaproxy >>>>/etc/init.d/mediaproxy start >>>>cd /usr/local/src/ >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you dont want >>>>Mediaproxy. I dont care, But remember SIP users wont be able to talk \n" >>>>fi >>>> >>>>echo "------ ser -------------------" >>>>printf "Do you want to INSTALL SER (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>## We make changes in the Makefile for all the modules, build our >>>>own tar and store it on our ftp server. >>>>#wget >>>><http://ftp.iptel.org/pub/ser/0.9.6/src/ser-0.9.6_src.tar.gz>http://ftp.iptel.org/pub/ser/0.9.6/src/ser-0.9.6_src.tar.gz >>>> >>>>cd /usr/local/src/ >>>>wget >>>><ftp://$TEMPLATE/configure_ser_server/ser-0.9.6_bingo.tar.gz>ftp://$TEMPLATE/configure_ser_server/ser-0.9.6_bingo.tar.gz >>>>echo "Press ENTER to start" >>>>read >>>>tar zxvf ser-0.9.6_bingo.tar.gz >>>>echo "Press ENTER to start" >>>>read >>>> >>>>cd ser-0.9.6 >>>>pwd >>>> >>>>#cd modules/acc >>>>#vi Makefile >>>>#remove comment from sql and radius lines >>>> >>>>#vi Makefile >>>>#remove *_radius from exclude list >>>> >>>>make all >>>>make install >>>>cd /usr/local/etc/ser/ >>>>wget >>>><ftp://$TEMPLATE/configure_ser_server/configfiles/ser.cfg>ftp://$TEMPLATE/configure_ser_server/configfiles/ser.cfg >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you don't want a >>>>softswitch you. Just playing with me HUHH, OK \n" >>>>fi >>>> >>>># >>>><http://www.openser.org/docs/openser-radius-1.0.x.html>http://www.openser.org/docs/openser-radius-1.0.x.html >>>># ---------------------------------------------- >>>>echo "------ APACHE -------------------" >>>>printf "Do you want to INSTALL APACHE (Y/N)?" >>>>read X >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><http://www.axint.net/apache/httpd/httpd-2.2.3.tar.gz>http://www.axint.net/apache/httpd/httpd-2.2.3.tar.gz >>>>tar zxvf httpd-2.2.3.tar.gz >>>>cd httpd-2.2.3 >>>>./configure --enable-shared=max --enable-module=rewrite >>>>--enable-module=so --with-apxs2 >>>>make >>>>make install >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you dont want APACHE. >>>>You might need this for serweb. \n" >>>>fi >>>> >>>>echo "------ IMAP -------------------" >>>>printf "Do you want to INSTALL IMAP, (Y/N)?" >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>>cd /usr/local/src/ >>>>#wget >>>><http://www.mirrorservice.org/sites/tp.cac.washington.edu/imap/old/imap-2004…>http://www.mirrorservice.org/sites/tp.cac.washington.edu/imap/old/imap-2004… >>>> >>>>wget >>>><ftp://192.168.2.201/configure_ser_server/imap-2004c1.tar.Z>ftp://192.168.2.201/configure_ser_server/imap-2004c1.tar.Z >>>> >>>>tar zxf imap-2004c1.tar.Z >>>> >>>>cd imap-2004c1 >>>>make slx SSLTYPE=none >>>>cd c-client/ >>>>cp c-client.a /usr/lib >>>>cp rfc822.h mail.h linkage.h /usr/include/ >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you dont want APACHE. >>>>You might need this for serweb. \n" >>>>fi >>>> >>>> >>>>echo "------ PHP 4.xx -------------------" >>>>printf "Do you want to INSTALL PHP 4.xx, If you dont install this >>>>then DONT complain if your serweb doesnt work (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>>wget >>>><http://museum.php.net/php4/php-4.3.10.tar.gz>http://museum.php.net/php4/php-4.3.10.tar.gz >>>>tar zxvf php-4.3.10.tar.gz >>>>cd php-4.3.10 >>>>./configure --with-apxs2=/usr/local/apache2/bin/apxs --with-mysql >>>>--with-imap >>>>make >>>>make install >>>> >>>>cp php.ini-dist /usr/local/lib/php.ini >>>>cd /usr/local/apache2/ >>>>wget >>>><ftp://$TEMPLATE/configure_ser_server/configfiles/httpd.conf>ftp://$TEMPLATE/configure_ser_server/configfiles/httpd.conf >>>> >>>> >>>>#vi /usr/local/apache2/conf/httpd.conf >>>># AddType application/x-httpd-php .php .phtml >>>># LoadModule php4_module libexec/libphp4.so >>>># >>>># modify the index.html line and add index.php to the list >>>># >>>># Listen 81 >>>> >>>>/usr/local/apache2/bin/apachectl start >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you refused to >>>>install PHP. You will need this for serweb. \n" >>>>fi >>>> >>>>echo "------ PHP 4.xx -------------------" >>>>printf "Do you want to INSTALL PHP 4.xx, If you dont install this >>>>then DONT complain if your serweb doesnt work (Y/N)?" >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>> >>>>wget >>>><http://ftp.iptel.org/pub/serweb/serweb-0.9.6.tar.gz>http://ftp.iptel.org/pub/serweb/serweb-0.9.6.tar.gz >>>> >>>>tar zxvf serweb-0.9.6.tar.gz >>>> >>>>mv serweb-0.9.6 serweb >>>># next few line have been taken care of httpd.conf in apache installation. >>>> >>>>#vi /usr/local/apache2/conf/httpd.conf >>>> >>>>#Alias /serweb "/usr/local/serweb/html/" >>>> >>>>#<Directory "/usr/local/serweb/html"> >>>># AllowOverride None >>>># Options None >>>># Order allow,deny >>>># Allow from all >>>>#</Directory> >>>> >>>>cd /usr/local/serweb/config >>>> >>>>#vi config* >>>> modify all config files are per your env. >>>> >>>>#edit: >>>>#vi /usr/local/etc/ser/ser.cfg >>>>#fifo_mode=666 >>>> >>>>#Send daily missed calls >>>>#Crontab line: >>>>#5 0 * * * /usr/local/bin/php >>>>/usr/local/serweb/scripts/cron_job/send_daily_missed_calls.php >>>> >>>>#Server monitoring >>>>#Crontab line: >>>>#* * * * * /usr/local/bin/php >>>>/usr/local/serweb/scripts/cron_job/read_ser_moni.php >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you refused to >>>>install PHP. You will need this for serweb. \n" >>>>fi >>>> >>>> >>>>echo "------ DIALIP_ADMIN -------------------" >>>>printf "Do you want to INSTALL DIALUP_ADMIN (Y/N)?" >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>>wget >>>><http://nchc.dl.sourceforge.net/sourceforge/dialup-admin/dialup_admin-1.62.t…>http://nchc.dl.sourceforge.net/sourceforge/dialup-admin/dialup_admin-1.62.t… >>>> >>>> >>>>tar zxvf dialup_admin- 1.62.tar.gz >>>>mv dialup_admin /usr/local/radiusadmin >>>> >>>>#vi /usr/local/apache2/conf/httpd.conf >>>> >>>>#Alias /radius "/usr/local/radiusadmin/htdocs/" >>>> >>>>#<Directory "/usr/local/radiusadmin/htdocs"> >>>># AllowOverride None >>>># Options None >>>># Order allow,deny >>>># Allow from all >>>>#</Directory> >>>> >>>>#cd /usr/local/radiusadmin/sql >>>># import all sql queries to radius database >>>>#cd /usr/local/radiusadmin/conf >>>># modify config files >>>> >>>> >>>>********************** >>>> >>>> >>>>On Dec 10, 2007 9:15 PM, ram >>>><<mailto:talk2ram@gmail.com>talk2ram(a)gmail.com > wrote: >>>> >>>>On Dec 10, 2007 11:02 PM, Jai Rangi >>>><<mailto:jprangi@gmail.com>jprangi(a)gmail.com> wrote: >>>>Preparing Kickstart file is a good idea. >>>>How about running a post installation script in the kickstart >>>>file. We have done this a lots on time in FC*. The process should >>>>be something like this, >>>>1. Build a default kickstart file that will install all the >>>>required packages for all the ser components. >>>>2. The kickstart file have an option to run a post installation script. >>>>3. Now we need to have one zip file of all the components we want >>>>to install related to ser. (ser, werbwer, sems, sipp, sppsak >>>>etc). That file can be placed either on the CDs or can be >>>>downloaded from somewhere. All the components can be packaged in >>>>the one zip file of they can be downloaded separately. >>>>4. The script can be interactive. >>>> >>>> >>>> >>>>yes that will be the idea, for the CD install >>>> >>>>after install of OS, we can run post install script to compile >>>>rest all required for >>>>SER ( or compile already for the same OS Distro and move them >>>>directly to the respective Directory) >>>> >>>> >>>> >>>> >>>>BTW I have a small postinstallation shell script like that, and I >>>>can share that. >>>> >>>> >>>>post to group, may be if require people may use. >>>> >>>>ram >>>> >>>> >>>>-Jai >>>> >>>> >>>>On Dec 10, 2007 3:42 AM, SIP <<mailto:sip@arcdiv.com> >>>>sip(a)arcdiv.com> wrote: >>>>Greger V. Teigre wrote: >>>> > [removed sems, semsdev, and serdev from cc] >>>> > >>>> > >>>> > SIP wrote: >>>> >> Something I noted about the project. While I am all for a >>>> virtual image >>>> >> for testing purposes and getting things up and running while we're >>>> >> creating this project, I think the ultimate goal should be something >>>> >> like an ISO that's ready to go on a server. There are many of >>>> us in the >>>> >> systems world who have not bought into the whole virtualisation >>>> >> marketing gimmick when it comes to system deployment, and for >>>> the casual >>>> >> home user who's less comfortable with using VMs for an actual >>>> production >>>> >> system (which is, to be honest, a lot of people), it might be >>>> easier in >>>> >> the long term to get everything the way we want it and then >>>> make an ISO >>>> >> that's ready to drop onto whatever machine with all the >>>> components we need. >>>> >> >>>> > Ok. When you say "ISO that's ready to drop onto", do you have a >>>> > particular procedure in mind? >>>> > >>>> > g-) >>>>Something like taking a kickstart file and making an installable livecd >>>>from it (something similar to the LiveCD berlios project). I don't KNOW >>>>of one that works to create installable ISOs in CentOS, but I'm sure >>>>there must be... somewhere... maybe....? >>>>N. >>>>_______________________________________________ >>>>Serusers mailing list >>>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>>_______________________________________________ >>>Serusers mailing list >>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >>_______________________________________________ >>Serusers mailing list >><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >>

1 0

[Serusers] SER Video call GXV3000
by FaberK 17 Dec '07

17 Dec '07

Hi to All, I got a SER 0.8.14 up and running since more than 2 years. It's perfect. Now I'm testing 2 Grandstream GXV3000. They are able to register and make calls, but nothing to do with Video calls. I've tryied them on Asterisk, and they work. Video calls are great. I've notice that they use port 8002 for the video and on SER I got the rtpproxy listening on 8000, maybe is this the problem or is something else? Thanks to all. Have a nice day. -- .:FaberK:.

4 4

[Serusers] SER 2.0 installation - SER does not start due to missing modules
by Strassburger, Patrick 17 Dec '07

17 Dec '07

Hello all, I am a newbie in SER and am failing since a few days installing SER 2.0. After several installation attempts I would like to ask you for some help. I did a "make group_include="standard mysql" all" but cannot see module mysql - where do I find those two modules and can I add them afterwards? Kind regards Patrick I get the following error and do not know how to resolve this: [root@ser ser]# ser -c 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: cannot open shared object file: No such file or directory 0(5613) parse error (55,13-14): failed to load module 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/xmlrpc.so>: /usr/local/lib/ser/modules/xmlrpc.so: cannot open shared object file: No such file or directory 0(5613) parse error (75,13-14): failed to load module 0(5613) set_mod_param_regex: No module matching <xmlrpc> found 0(5613) parse error (142,20-21): Can't set module parameter 0(5613) parse error (235,16-17): unknown command, missing loadmodule? ERROR: bad config file (4 errors) [root@ser ser]# ls -l /usr/local/lib/ser/modules/ total 4944 -rwxr-xr-x 1 root root 67012 Dec 17 15:02 acc_db.so -rwxr-xr-x 1 root root 73256 Dec 17 15:02 acc_syslog.so -rwxr-xr-x 1 root root 69954 Dec 17 15:02 auth_db.so -rwxr-xr-x 1 root root 58083 Dec 17 15:02 auth.so -rwxr-xr-x 1 root root 37353 Dec 17 15:02 avp_db.so -rwxr-xr-x 1 root root 168776 Dec 17 15:02 avpops.so -rwxr-xr-x 1 root root 79122 Dec 17 15:02 avp.so -rwxr-xr-x 1 root root 279393 Dec 17 15:02 ctl.so -rwxr-xr-x 1 root root 72908 Dec 17 15:02 db_ops.so -rwxr-xr-x 1 root root 63015 Dec 17 15:02 dispatcher.so -rwxr-xr-x 1 root root 21612 Dec 17 15:02 diversion.so -rwxr-xr-x 1 root root 114544 Dec 17 15:02 domain.so -rwxr-xr-x 1 root root 45712 Dec 17 15:02 enum.so -rwxr-xr-x 1 root root 84369 Dec 17 15:02 exec.so -rwxr-xr-x 1 root root 81406 Dec 17 15:02 fifo.so -rwxr-xr-x 1 root root 53450 Dec 17 15:02 flatstore.so -rwxr-xr-x 1 root root 43426 Dec 17 15:02 gflags.so -rwxr-xr-x 1 root root 87582 Dec 17 15:02 lcr.so -rwxr-xr-x 1 root root 35891 Dec 17 15:02 maxfwd.so -rwxr-xr-x 1 root root 112643 Dec 17 15:02 mediaproxy.so -rwxr-xr-x 1 root root 87493 Dec 17 15:02 msilo.so -rwxr-xr-x 1 root root 187830 Dec 17 15:02 nathelper.so -rwxr-xr-x 1 root root 21211 Dec 17 15:02 options.so -rwxr-xr-x 1 root root 131901 Dec 17 15:02 pdt.so -rwxr-xr-x 1 root root 272121 Dec 17 15:02 permissions.so -rwxr-xr-x 1 root root 122166 Dec 17 15:02 pike.so -rwxr-xr-x 1 root root 26180 Dec 17 15:02 print.so -rwxr-xr-x 1 root root 47605 Dec 17 15:02 ratelimit.so -rwxr-xr-x 1 root root 152851 Dec 17 15:02 registrar.so -rwxr-xr-x 1 root root 114904 Dec 17 15:02 rr.so -rwxr-xr-x 1 root root 63288 Dec 17 15:02 sanity.so -rwxr-xr-x 1 root root 66950 Dec 17 15:02 sl.so -rwxr-xr-x 1 root root 34423 Dec 17 15:02 speeddial.so -rwxr-xr-x 1 root root 120815 Dec 17 15:02 textops.so -rwxr-xr-x 1 root root 43284 Dec 17 15:02 timer.so -rwxr-xr-x 1 root root 848970 Dec 17 15:02 tm.so -rwxr-xr-x 1 root root 117605 Dec 17 15:02 uac.so -rwxr-xr-x 1 root root 38801 Dec 17 15:02 unixsock.so -rwxr-xr-x 1 root root 38987 Dec 17 15:02 uri_db.so -rwxr-xr-x 1 root root 38611 Dec 17 15:02 uri.so -rwxr-xr-x 1 root root 351480 Dec 17 15:02 usrloc.so -rwxr-xr-x 1 root root 100362 Dec 17 15:02 xlog.so [root@ser ser]# ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900.

2 1

[OpenSER-Users] IMC Module
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

Hi All, I am using openser as a proxy and presence server. now i am add new module IMC for support chat conference. But how i can test changes are right???????? There is any standard phone Or sip client which support IMC module and able to send messages like create,join.... for conference. So, i can test our sip proxy........... Thanks & Regards, --Amit

2 6

[OpenSER-Users] OpenSER Oracle
by Martin.Gruber1＠infineon.com 17 Dec '07

17 Dec '07

HI, How did you solve the problem....Suddenly I also got an error: [S1000][unixODBC][Oracle][ODBC][Ora]ORA-12514: TNS:listener does not currently know of service requested in connect descriptor Does anybody know the problem? Regards Martin

2 1

[OpenSER-Users] xmpp error
by TCB 17 Dec '07

17 Dec '07

i'm trying to setup openser-1.3 with xmpp support . I get the following error when I to IM another user. :cmd_send_message: invalid content-type 0xff00f -- TC

2 1

[OpenSER-Users] Help regarding tls
by Vijender Singh, Noida 17 Dec '07

17 Dec '07

Hi Sir, For enabling tls support on openser version 1.0.1 on Fedora 7, I have the attached openser.cfg file which I am using. tls_certificate = "/sip-server/tls/tools/user/user/user-cert.pem" tls_private_key = "/sip-server/tls/tools/user/user-privkey.pem" tls_ca_list = "/sip-server/tls/tools/user/user-calist.pem" The three given certificates are generated using scripts gen_rootCA.sh and gen_usercert.sh. When I try to run openser, it gives me the error saying "ERROR: bad config file (4 errors)" I have done configuration on the basis of the instruction given in the link http://www.openser.org/docs/tls.html#AEN93. I also installed the openssl-0.9.8g on my Fedora machine. Can you help me out as I need to complete my work on very high priority? Thanks in advance Vijender DISCLAIMER: ----------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. Any views or opinions presented in this email are solely those of the author and may not necessarily reflect the opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of the author of this e-mail is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any mail and attachments please check them for viruses and defect. -----------------------------------------------------------------------------------------------------------------------

2 1

[OpenSER-Users] Stored Procedure
by Il Neofita 17 Dec '07

17 Dec '07

Hi, I am looking to openser 1.3 and I cannot figure out how to use the mysql stored procedure. Anyone, can post a little example?

1 0

[OpenSER-Users] multi-domain support
by satish patel 17 Dec '07

17 Dec '07

This is my config file debug=3 #fork=no log_stderror=yes listen=192.168.104.20 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("domain", "db_url", "mysql://root:tulip@localhost/ser") #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) modparam("domain", "db_mode", 1) # --- main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (is_uri_host_local() || uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (is_uri_host_local() || uri==myself) { if (method=="REGISTER") { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (is_from_local() || from_uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (is_uri_host_local() || uri==myself) { if (method=="REGISTER") { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } when i run ser i got error modules error . is it correct or not can u check it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[OpenSER-Users] [LCR] how does "to_gw()" work?
by Iñaki Baz Castillo 17 Dec '07

17 Dec '07

Hi, I read in LCR doc: 1.4.4. to_gw([group-id]) Checks if in-dialog request goes to a gateway. If an optional group-id is given, only gateways belonging to this group are checked. I suppose that a gateway uses its scheme:IP:port as "Contact" in 200 OK, so "to_gw()" function matches that scheme:IP:port in RURI and verifies if there is a To tag to match an in-dialog message and deliver it to gateway. Is it? Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 3

[OpenSER-Users] [Fwd: Re: IMC Module]
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

1 0

[OpenSER-Users] Any linux/windows/mac os SIPs client to use with OpenSER-tls ?
by Marc LEURENT 17 Dec '07

17 Dec '07

Good afternoon, Have you got names of SIP client with sip-tls support that you have seen working with OpenSER? Thanks I'm asking that before having to deploy a WPA2-Enterprise wireless network to ensure masked signaling / (and perhaps masked communications)

2 1

[Serusers] multi-domain support
by satish patel 17 Dec '07

17 Dec '07

Dear all i have going to install ser with multidomain support like aaa(a)example.com bbb(a)openser.org ccc(a)iptel.org Something like this but i done know how to configure it in CFG file is there anybody have runing configuration or sample example file plz hint me and send it me .... $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Chat on a cool, new interface. No download required. Click here.

2 1

[OpenSER-Users] Allow_trusted problems
by CSB 17 Dec '07

17 Dec '07

For some reason I cannot get allow_trusted() to evaluate to true for a matching URI. Relevant sections of the config (OpenSER 1.1): modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", "mysql://username:password@localhost/ser") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") route { if (!allow_trusted()) { xlog("!allow_trusted. rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); }; } The trusted table contains the following: +----------------+-------+--------------+------+ | src_ip | proto | from_pattern | tag | +----------------+-------+--------------+------+ | 202.200.20.200 | any | NULL | NULL | +----------------+-------+--------------+------+ 1 row in set (0.00 sec) I can see that the following query is being executed: select proto,from_pattern,tag from trusted where src_ip='202.200.20.200' But no match is made because the log file shows: Dec 17 15:37:56 beta /usr/local/sbin/openser[7209]: !allow_trusted. rs <null> si 147.200.20.200 rm INVITE ru sip:5551234@147.200.20.200 tu sip:5551234@147.200.20.200 fu sip:021446202@202.200.20.200 rr <null> Could anyone provide any suggestions on what I'm doing wrong? Regards Cameron

2 5

[OpenSER-Users] [Fwd: IMC Module]
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

1 0

[OpenSER-Users] OpenSER billing options
by Andy Smith 17 Dec '07

17 Dec '07

Hi list, are there alternatives to Radius for billing integration with OpenSER? The problem Radius seems to pose me is that it is per user and therefore requires to have a full database of every softphone/SNOM/etc user, which is admin intensive. Does anyone know if there is a method of accounting data per SIP peer (for example per Asterisk system attached to OpenSER) which could record call details without needing to know about all the actual handset users? I suppose you can do anything you want with the accounting module so this is really a "does anyone already do this" question so I dont need to invent everything from scratch... thanks for any suggestions, Andy.

2 1

[OpenSER-Users] Problem Registering via TCP
by Krom 16 Dec '07

16 Dec '07

I have a UA that I am attempting to register to OpenSER 1.1.1 via TCP and the only hint I can give it the following statement that is placed in the syslog file. WARNING: io_wait_loop_sigio_rt: ignoring event 41 on fd 0 (fm->fd=0, fm->data=(nil)) I have placed in the first line of the main route the following just to see if the registration is making it to the main route and it is not. if(proto==TCP) { xlog("L_INFO","SIP message was received over TCP\n"); }; I can tell from tcpdump that the packet is being received and a complete SYN,ACK is successful. Global Config: listen=10.1.1.28:5060 disable_tls=1 tcp_children=4 tcp_send_timeout=3 tcp_connect_timeout=5 tcp_connection_lifetime=3600 A netsat shows the port in a listening state. When starting OpenSER it shows that it is listening on TCP:IP_ADDR:5060 When the UA sends a register request, a connection is established, the one statement above is placed in the log file and then the connection goes to a CLOSE_WAIT state. Any suggestions. I know at least one of us has seen this before. I searched the list archives and found one thread in Nov 06 discussing the same issue; but no follow-up of resolution posted. http://lists.openser.org/pipermail/users/2006-November/007653.html -- -- Krom kyle.romulas(a)gmail.com

1 1

[OpenSER-Users] CPL doesn't allow "redirect" to url="http://webpage"
by Iñaki Baz Castillo 16 Dec '07

16 Dec '07

Hi, I want to set a redirect to a web page in CPL but I get error: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> <notfound> <location url="http://google.es"> <redirect /> </location> </notfound> </lookup> </incoming> </cpl> ~# openserctl fifo LOAD_CPL sip:user@domain /etc/openser/cpl/user(a)domain.xml 500 Bad CPL file Log:: Error: Encoding of the CPL script failed It works if I set (as usual): <location url="sip:user2@domain"> Why doesn't CPL module allow redirection to non SIP target? -- Iñaki Baz Castillo

1 0

[OpenSER-Users] OpenSER with multidomain support
by satish patel 15 Dec '07

15 Dec '07

Dear all I have post many question about multidomain support help for openser i need cfg file which you ppl are useing caz openser.cfg is bit harder for me and i have also find many help on ser on google but still i m confuse about parmeter and options if anybody have done this configuration send it me or explain me how to do it ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

2 1

[OpenSER-Users] get Count of items in lists
by Dave Singer 14 Dec '07

14 Dec '07

using openser 1.3, is there a way to get a count of how many items there are in an avp variable other than: while ( is_avp_set("$avp(s:fork_uri)") ) { $var(fork_uri_count) = $var(fork_uri_count) + 1 ; avp_copy("$avp(s:fork_uri)", "$avp(s:fork_uri_rev_order)/d") ; } then if you wanted things in the same order as they were originally you would also have to: while ( is_avp_set("$avp(s:fork_uri_rev_order)") ) { avp_copy("$avp(s:fork_uri_rev_order)", "$avp(s:fork_uri)/d") ; } talk about an expensive counting Several ways I could see this implimented: New function avp_count("$avp(s:myvar)", "$avp(s:count_result)") or avp_count("$avp(s:myvar)") where $rc contains the result. is_avp_set("$avp(s:myvar)") where $rc is set to the count, probably best as you could test if exitst and the go right to processing. avp_check("$avp(s:myvar)", "cn/g") where $rc is set to the count. It could always set $rc to count of found matches for when the g flag is used. Of course using $rc for is_avp_set() or avp_check() depends on if setting that to non-zero makes a test like if( is_avp_set("$avp(s:myvar)") ) always trigger as false. also on getting a count ... another place I need to get a count is how many times a character(s) (or shoot the moon, a regex) occurs in a string. for handeling things like transformations like {s.select,index,seperator}, eg: $(var(findme){ s.select,$var(count),,}) sidenote ... (not sure what values for seperator are valid) a transformation like {s.count,seperator} Here is an example of something I was trying to do that would have benifited from knowing the count of seperator. I think that I got a warning or error when the test for the while indexed past the end of the string. xlog("L_INFO", "Original findme list to parse: $avp(s:findme)\n"); if ( $avp(s:findme) =~ "\|" ) { $var(findme) = $(avp(s:findme){s.select,0,|}) ; $var(findme_tm_out) = $(avp(s:findme){s.select,1,|}) ; xlog("L_INFO", "This findme has a timeout: $var(findme_tm_out)\n"); } else { $var(findme) = $avp(s:findme) ; } xlog("L_INFO", "findme list to parse: $var(findme)\n"); $var(findme_cnt) = 0 ; while ($(var(findme){s.select,$var(findme_cnt),,}{s.len }) > 0 ) { xlog("L_INFO", "Loop $var(findme_cnt), Adding $(var(findme){s.select,$var(findme_cnt),,}) to TO_list\n"); $avp(s:TO_list) = $(var(findme){s.select,$var(findme_cnt),,}) ; $var(findme_cnt) = $var(findme_cnt) + 1 ; } can/do $var()'s stack like $avp()'s? Thanks Dave

2 4

Re: [OpenSER-Users] Missed calls (I?aki Baz Castillo)
by David Villasmil Govea 14 Dec '07

14 Dec '07

te refieres a ésto? ya está en el openser.cfg if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; > > > El Friday 14 December 2007 13:55:01 David Villasmil Govea escribi?: > > * Furthermore,?Openser doesn't see all INVITES+BYE, so i can't make > cdrs!! > > I've read somewhere that to log all transaction messages i should have > > "strict route" as oppossed to "loose route". > > No, you need to add "Record-Route" header. This is done by > "record_route()" > function of rr module. > > -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Re: [OpenSER-Users] Users Digest, Vol 31, Issue 46
by David Villasmil Govea 14 Dec '07

14 Dec '07

te refieres a ésto? ya está en el openser.cfg if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; > > > El Friday 14 December 2007 13:55:01 David Villasmil Govea escribi?: > > * Furthermore,?Openser doesn't see all INVITES+BYE, so i can't make > cdrs!! > > I've read somewhere that to log all transaction messages i should have > > "strict route" as oppossed to "loose route". > > No, you need to add "Record-Route" header. This is done by > "record_route()" > function of rr module. > > -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

[OpenSER-Users] openser config issue or asterisk bug
by Patrick Baker 14 Dec '07

14 Dec '07

Internal UA's connecting to the sip proxy work fine, but as soon an External UA connects to the sip proxy via a NAT'd public IP address - the call lasts for approx 20 seconds then disconnects. From what I can see openser is attempting to respond to asterisk after the session is initiated via the public IP NAT'd address - and is unable since asterisk and openser are behind the firewall on the internal network ( ELBOW routing ). Any help getting this working would be most appreciated, I would also like to document the issue to help others that encounter similar problems as mine. External phones: UA -> [(public IP) Firewall (NAT to localip)] -> openser -> asterisk Internal phones: UA -> openser -> asterisk #### Layout ( asterisk and openser are on the same machine ) NATD public ip = xxx.206.xxx.136 -> 10.3.1.33 phone ip = xxx.206.xxx.137 openser = 10.3.1.33 p5060 asteriskSIP = 10.3.1.34 p5060 #### Asterisk SIP Config [general] canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5060 bindaddr=10.3.1.34 sipdebug=yes [openser] type=friend context=default insecure=very externalnotify=yes allow=all host=10.3.1.33 #### SER LOG Dec 13 22:15:03 phonesys-slave openser[15976]: New request - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Callee is not local - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.34:5060;transport=udp F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15973]: Reply - S=100 D=Trying F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15978]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:04 phonesys-slave openser[15989]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:05 phonesys-slave openser[15986]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:07 phonesys-slave openser[15976]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 #### Asterisk LOG *CLI> [Dec 13 22:15:03] <--- SIP read from 10.3.1.33:5060 ---> INVITE sip:500@10.3.1.34:5060;transport=udp SIP/2.0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone> CSeq: 2 INVITE Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Contact: <sip:pbaker@xxx.206.xxx.137> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 69 Content-Type: application/sdp Content-Length: 254 v=0 o=- 978310717 978310717 IN IP4 xxx.206.xxx.137 s=Polycom IP Phone c=IN IP4 xxx.206.xxx.137 t=0 0 m=audio 35022 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Dec 13 22:15:03] --- (16 headers 11 lines) --- [Dec 13 22:15:03] == Using TOS bits 0 [Dec 13 22:15:03] == Using CoS mark 5 [Dec 13 22:15:03] Sending to 10.3.1.33 : 5060 (no NAT) [Dec 13 22:15:03] Using INVITE request as basis request - f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 [Dec 13 22:15:03] No user 'pbaker' in SIP users list [Dec 13 22:15:03] Found peer 'openser' for 'pbaker' from 10.3.1.33:5060 [Dec 13 22:15:03] Found RTP audio format 0 [Dec 13 22:15:03] Found RTP audio format 8 [Dec 13 22:15:03] Found RTP audio format 18 [Dec 13 22:15:03] Found RTP audio format 101 [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Found audio description format PCMU for ID 0 [Dec 13 22:15:03] Found audio description format PCMA for ID 8 [Dec 13 22:15:03] Found audio description format G729 for ID 18 [Dec 13 22:15:03] Found audio description format telephone-event for ID 101 [Dec 13 22:15:03] Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) [Dec 13 22:15:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Looking for 500 in default (domain 10.3.1.34) [Dec 13 22:15:03] list_route: hop: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> [Dec 13 22:15:03] <--- Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone> Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Length: 0 <------------> [Dec 13 22:15:03] -- Executing [500@default:1] Playback("SIP/xxx.206.xxx.136-090b5380", "demo-abouttotry") in new stack [Dec 13 22:15:03] Audio is at 10.3.1.34 port 47068 [Dec 13 22:15:03] Adding codec 0x4 (ulaw) to SDP [Dec 13 22:15:03] Adding codec 0x8 (alaw) to SDP [Dec 13 22:15:03] Adding codec 0x100 (g729) to SDP [Dec 13 22:15:03] Adding non-codec 0x1 (telephone-event) to SDP [Dec 13 22:15:03] <--- Reliably Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 13 22:15:04] -- <SIP/xxx.206.xxx.136-090b5380> Playing 'demo-abouttotry.slin' (language 'en') [Dec 13 22:15:04] Retransmitting #1 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:05] Retransmitting #2 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:07] Retransmitting #3 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:11] Retransmitting #4 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:15] Retransmitting #5 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:18] -- Executing [500@default:2] Dial("SIP/xxx.206.xxx.136-090b5380", "IAX2/guest@misery.digium.com/s@default") in new stack [Dec 13 22:15:18] -- Called guest@misery.digium.com/s@default [Dec 13 22:15:18] -- Call accepted by 216.207.245.8 (format gsm) [Dec 13 22:15:18] -- Format for call is gsm [Dec 13 22:15:18] -- IAX2/216.207.245.8:4569-1 is ringing [Dec 13 22:15:19] -- IAX2/216.207.245.8:4569-1 answered SIP/xxx.206.xxx.136-090b5380 [Dec 13 22:15:19] Retransmitting #6 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2337 retrans_pkt: Maximum retries exceeded on transmission f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 for seqno 2 (Critical Response) [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2364 retrans_pkt: Hanging up call f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 - no reply to our critical packet. [Dec 13 22:15:23] -- Hungup 'IAX2/216.207.245.8:4569-1' [Dec 13 22:15:23] == Spawn extension (default, 500, 2) exited non-zero on 'SIP/xxx.206.xxx.136-090b5380' [Dec 13 22:15:24] Really destroying SIP dialog 'f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137' Method: INVITE #### PCAP Diagram |Time | xxx.206.xxx.137 | 10.3.1.33 | |5.056 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.058 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.058 | Status: 407 Proxy A |SIP: Status: 407 Proxy Authentication Required | |(5060) <------------------ (5060) | |5.140 | Request: ACK sip:50 |SIP: Request: ACK sip:500@xxx.206.xxx.136:5060 | |(5060) ------------------> (5060) | |5.165 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.165 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |6.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |7.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |9.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |13.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |17.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |21.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | #### OpenSER Config 1.2 ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Thu Dec 13 22:43:43 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Thu Dec 13 22:43:43 +0100 2007' ######################################################################## listen = udp:10.3.1.33:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = yes check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) { xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } t_on_failure("1"); route(3); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); }

2 1

[OpenSER-Users] Missed calls
by David Villasmil Govea 14 Dec '07

14 Dec '07

Hello all, I've configured Openser 1.2 with accounting + mysql. at the momento i'm only loggin: BYE CANCEL INVITE REGISTER * How do i identify missed calls? * Furthermore, Openser doesn't see all INVITES+BYE, so i can't make cdrs!! I've read somewhere that to log all transaction messages i should have "strict route" as oppossed to "loose route". Is that enabled by: modparam("rr", "enable_full_lr", 0) ?? Thanks a lot! this is my config: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=20 disable_dns_blacklist=true # # --- module loading mpath="/usr/lib/openser/modules/" 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 "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" #loadmodule "domain.so" loadmodule "xlog.so" loadmodule "/usr/lib/openser/modules/acc.so" modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_url", "mysql://user:password@localhost/openser") modparam("acc", "report_ack", 0) modparam("acc", "db_missed_flag", 3) modparam("acc", "failed_transaction_flag", 4) modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # # CONSTANTS # # --- setting module parameters #modparam("domain","db_url","mysql://user:password@localhost/openser") #modparam("domain", "db_mode", 1) #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") modparam("xlog", "buf_size", 8192) # --- setting module parameters modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc|auth_db","db_url","mysql://user:password@localhost /openser") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "nat_bflag", 6) #modparam("usrloc", "use_domain", 0) # -- registrar params -- modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("auth_db", "use_domain", 0) # -- rr params -- #modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:1.2.3.5:8899") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@sip2.telmaxi.com.br") #modparam("registrar", "max_contacts", 1) # --- main routing logic route{ if (method=="INVITE") { xlog("L_NOTICE", "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(2); setflag(3); setflag(4); }; if (method=="BYE") { xlog ("L_NOTICE", "BYE - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="CANCEL") { xlog ("L_NOTICE", "CANCEL - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="REGISTER") { setflag(2); setflag(3); setflag(4); }; if (!mf_process_maxfwd_header("100")) { sl_send_reply("483","Too Many Hops"); exit; }; #if (msg:len >= 2048 ) { # sl_send_reply("513", "Message too big"); # exit; #}; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # if (!is_uri_host_local() ) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_NOTICE", "\nNew Registration\n"); if (!www_authorize("1.2.3.4", "subscriber")) { xlog("L_NOTICE", "\nRegistration FAILED! CLIENT: $au\n"); www_challenge("1.2.3.4", "0"); exit; }; if (isflagset(5)) { xlog("L_NOTICE", "\nRegistration OK\n"); setbflag(6); # if you want OPTIONS natpings uncomment next # setbflag(7); }; save("location"); exit; }; if (uri=~"^sip:0"){ xlog("L_NOTICE", "\n************** GOT OUTSIDE CALL ************\n"); ds_select_dst("1","2"); sl_send_reply("100","Trying"); forward(); exit(); } if (uri=~"^sip:7"){ ds_select_dst("1","2"); sl_send_reply("100","Trying"); forward(); exit(); } if (uri=~"^sip:1"){ ds_select_dst("1","2"); sl_send_reply("100","Trying"); forward(); exit(); } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); #}; }; route(1); } route[1] { xlog("L_NOTICE", "\n**************IN ROUTE 1************\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; }

2 1

[OpenSER-Users] Security hole in REGISTER's Contact using domain
by Iñaki Baz Castillo 14 Dec '07

14 Dec '07

Hi, Permissions module tries to avoid REGISTER with privileged IP's in Contact (using "register.deny" file) but I have some doubs about this security. I'll play with the example explained in "register.deny" file: --------------------------------------------------------------------------------------- # Suppose that we have a PSTN gateway with IP address 1.2.3.4 # We should prevent REGISTER messages that contain that IP # address in Contact header field because that can cause serious # security hole (a malicious user might be able to register such # a contact and bypass security checks performed by the SIP proxy). # # The following line prevents registering Contacts with IP 1.2.3.4 # (Don't forget to list also all hostnames that can be used to # reach the PSTN gateway) ALL : "^sip:.*1\.2\.3\.4" --------------------------------------------------------------------------------------- Ok, now a malicious user could just use SipSak to send a malicious REGISTER to call for free to a PSTN number 01666555444: ~# sipsak -U -C sip:01666555444@1.2.3.00004 -a passwd -s sip:200@domain.org Note the "000004" !!!! So this causes a entry in "location" with fields: - username = 200 - domain = domain.org - contact = sip:01666555444@1.2.3.00004 And sure 1.2.3.00004 is a valid IPv4. This is: if the user calls itself (sip:200@domain.org) he'll get a free PSTN call. Oppss... Ok, a solution could be to improve the regular expression by avoiding any number of 0's: ALL : "^sip:.*0*1\.0*2\.0*3\.0*4" Ok, but now the malicious user can register a domain "hacking_my_proxy.com" to resolve to IP 1.2.3.4, and send this REGISTER: ~# sipsak -U -C sip:01666555444@hacking_my_proxy.com -a passwd -s sip:200@domain.org So this will bypass the "register.deny" policy !!!! Note that "register.deny" file says: # (Don't forget to list also all hostnames that can be used to # reach the PSTN gateway) Of course, it's not possible to list all hostnames and domain resolving an IP (anyone can register a domain to any IP). So then... is it really valid this "register.deny" security???? Solution for this? ------------------------- - Forbid hostnames or domains in Contact: Ohh, too much anti-RFC 3261 (what would "alice(a)pc33.atlanta.com" think about it? XDDD). - Do a DNS query for the "Contact" during REGISTER: What about if DNS changes later? - Match the resolved IP against IP's in "register.deny" for every INVITE leaving OpenSer. Humm. - Avoid OpenSer using internet DNS system (so "hacking_my_proxy.com" wouldn't be resolved) and allow just secure domains (internal DNS or /etc/hosts): and what about outbound calls? isn't this solution an atrocity? How to handle it? is it not a real security hole? Comments are welcome. Regards. -- Iñaki Baz Castillo

5 13

[OpenSER-Users] which media server to use to write a Ringback Tone application
by Srinivas Kotamarti 14 Dec '07

14 Dec '07

All, I am trying to write a carrier grade Ringback Tone application on top of OpenSER, does anybody know which media server (preferably OpenSource) to use? Is WeSIP the only way to write an application on top of OpenSER or there are otherways? Also, does anybody know how the applications in WeSIP perform can they be deployed as a carrier grade solution? Regards --Srinivas Kotamarti

2 1

[OpenSER-Users] 1.3.0 FreeBSD port committed to svn
by Jesus Rodriguez 14 Dec '07

14 Dec '07

Hi, Yesterday night i committed to the svn the update of the FreeBSD port to 1.3.0 version. This update will also be committed in the FreeBSD ports tree as soon as possible. Meanwhile, if you want to use the port or create your own packages, you can obtain the port here: http://openser.svn.sourceforge.net/viewvc/openser/branches/1.3/packaging/fr… Regards. Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

1 0

[OpenSER-Users] PARALLEL FORKING
by Gomtesh Jain 14 Dec '07

14 Dec '07

Hi All , I am trying to do parallel forking from Openser.But when I call then one user gets 2 calls and rest of the users get one call . Please any one explain why it is happening. Here I attach my openser.cfg file... # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls: domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("192.168.3.15", "subscriber")) { # www_challenge("192.168.3.15", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); log("I M IN lookup"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } else { append_branch(); append_hf("P-hint: usrloc applied\r\n"); t_relay(); } }; route(1); }

2 1

Re: [OpenSER-Users] Preparing the release of OpenSER 1.3 "BlackCat"
by Sergio Gutierrez 14 Dec '07

14 Dec '07

Hi Bogdan. The compilation on Solaris 10/SPARC has ended succesfully. Acording to my current setup, in particular, related to fulfilled dependencies, my final gmake command was: *gmake exclude_modules="jabber mi_xmlrpc osp perl postgres unixodbc snmpstats xmpp tlsops db_berkeley ldap mi_dm perlvdb xcap_client" *I have mysql 5.0.22 and radiusclient 0.5.2 The attached file contains the whole warnings reported at the compilation process; most of them are related to assignments without casting, and implicit declaration of functions. I will peform the TLS compilation, and the testing on Solaris x86, and I will be notifying you as soon as they end. If there is something I can help, or another test you would like I perform, please let me know. Best Regards. Sergio Gutiérrez EPM Telecomunicaciones Medellin, Colombia Tel: +574 - 3802947 On Dec 12, 2007 4:54 PM, Sergio Gutierrez <saguti(a)gmail.com> wrote: > Hi Bogdan. > > Continuing with compilation, this error appeared when compiling LDAP > module: > > In file included from ldap_api_fn.c:40: > ldap_api_fn.h:57: error: conflicting types for 'ldap_url_search' > /usr/include/ldap.h:789: error: previous declaration of 'ldap_url_search' > was here > ldap_api_fn.h:57: error: conflicting types for 'ldap_url_search' > /usr/include/ldap.h:789: error: previous declaration of 'ldap_url_search' > was here > ldap_api_fn.c: In function `ldap_params_search': > ldap_api_fn.c:211: warning: implicit declaration of function > `LDAP_API_ERROR' > > In my case, I do not have OpenLDAP libraries installed, but my system has > the LDAP library which is standard on Solaris since Solaris 9. > > checking at the manual, I see the declaration on Solaris is: > > int *ldap_url_search*(LDAP *ld, char *url, int attrsonly); > > > Please, let me know how can we continue. > > Regards. > > Sergio. > > > > On Dec 12, 2007 4:22 PM, Bogdan-Andrei Iancu < bogdan(a)voice-system.ro> > wrote: > > > Hi Sergio, > > > > Thanks for the feedback - I'm glad we were able to catch this issue in > > last minutes before release :). I also applied your patch for the > > Makefile of the carrierroute module. > > > > Regards, > > Bogdan > > > > Sergio Gutierrez wrote: > > > Hi Bogdan. > > > > > > The compilation so far is going right, with no warnings; > > > > > > A modification I did is I installed libconfuse, because for our > > > particular setup we are interested in exploring the carrierroute > > > module; I installed at my LOCALBASE, and at compiling the module, it > > > failed; trying to fix it I added a couple lines to the Makefile of the > > > module as indicated at the following diff; I am not sure if they are > > > required, but they fixed my problem: > > > > > > > > > ------------------------- BEGIN DIFF > > > --- Makefile Wed Dec 12 16:05:06 2007 > > > +++ Makefile.saguti Wed Dec 12 16:04:58 2007 > > > @@ -8,7 +8,8 @@ > > > include ../../Makefile.defs > > > auto_gen= > > > NAME=carrierroute.so > > > -LIBS=-lconfuse > > > +LIBS+=-L$(LOCALBASE)/lib -lconfuse > > > +DEFS+=-I$(LOCALBASE)/include > > > > > > include ../../Makefile.modules > > > > > > ------------------------ END DIFF > > > > > > I will try again the compilation, and I hope to test it too on Solaris > > > x86. > > > > > > Best regards. > > > > > > Sergio. > > > > > > > >

2 4

[OpenSER-Users] OpenSER 1.3.0 BlackCat is released !
by Bogdan-Andrei Iancu 14 Dec '07

14 Dec '07

Hi, I'm glad to announce that OpenSER 1.3.0 was released - I want to thanks to all people that contributed to it, as code, documentation, assistance, packaging, etc... The SVN branch for 1.3 and tag for 1.3.0 was created, so feel free to download : # svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 openser As tarballs, only source (tls and non-tls) and linux binaries are available on openser.org site. In the next following days packages will be uploaded on both openser.org and SF sites. There are still pending to upload on the sites a news about the features in 1.3.0, about migration from 1.2 to 1.3 - all these will be done in the following days. More announcement email (for detailed) will follow ;). Regards, Bogdan

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[OpenSER-Users] SUSE packages
by Daniel-Constantin Mierla 13 Dec '07

13 Dec '07

Thanks to Peter Nixon, the OpenSER 1.3.0 packages for different flavors of SUSE are already available: http://download.opensuse.org/repositories/network:/telephony/ More details about how to use the repository with different package managers: http://www.openser.org/mos/view/Download/ Cheers, Daniel

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[OpenSER-Users] openser 1.3.0 release notes
by Daniel-Constantin Mierla 13 Dec '07

13 Dec '07

Hello, the release notes for version 1.3.0 are available at: http://www.openser.org/mos/view/OpenSER-v1.3.x-Release-Notes/ If you feel that something is missing, let me know to update the page. Enjoy the new version! Daniel

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[Serusers] loose_ route domain import
by Jan Ruzicka 13 Dec '07

13 Dec '07

Hi, does loose_route import the domains from domain module? My multidomain 0.9.6 seems no to do this and I found same issues for openser (http://www.mail-archive.com/users@openser.org/msg12794.html). What about 2.x? I'm afraid the only workaround, as mentioned, is to use alias=domain for all domains served by server. Thanks Jan -- ------------------------------------------------------------------- Ing. Jan Ruzicka CESNET z. s. p. o. tel: +420234680499 Zikova 4 e-mail,sip: janru(a)cesnet.cz 160 00 Praha 6 icq:151316768 Czech Republic -------------------------------------------------------------------

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Re: [OpenSER-Users] OpenSER as SIP Proxy for PSTN Gateway
by Andy Smith 13 Dec '07

13 Dec '07

>Yes, define a peer in sip.conf: > >[openser] >host=XX.XX.XX.XX >insecure=invite > >In this way Asterisk allow calls from OpenSer without forcing them to >authenticate. Authenticacion had already place in OpenSer.So I guess this would function for calls coming from the PSTN gateway, what about for callsoriginating from SIP devices connected to Asterisk? Can I do some similar config on OpenSERso that calls authenticated on Asterisk dont require authentication for an INVITE on OpenSER?

2 1

[Serusers] SER authentication through FreeRadius with an LDAPbackend
by Daniel Serrano 13 Dec '07

13 Dec '07

Dear Greger V. Teigre, Did you find another solution for this issue instead of generate the new attribute in the LDAP DB? Some kind of workaround will be great!.. Thanks in advance. Regards, Daniel Serrano.

2 4

[Serusers] PARALLEL FORKING
by Gomtesh Jain 13 Dec '07

13 Dec '07

Hi All , I am trying to do parallel forking from SER.But when I call then one user gets 2 calls and rest of the users get one call . Please any one explain why it is happening. Here I attach my openser.cfg file... # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("192.168.3.15", "subscriber")) { # www_challenge("192.168.3.15", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); log("I M IN lookup"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } else { append_branch(); append_hf("P-hint: usrloc applied\r\n"); t_relay(); } }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP #if (!t_relay()) { if (!forward()) { sl_reply_error(); }; exit; } Gomtesh

2 1

[Serusers] Can SER be used as a SIP A/S
by Srinivas Kotamarti 13 Dec '07

13 Dec '07

All, Can SER be used as a SIP A/S for example to write a carrier grade Ringback Tone application on top of SER. Does anybody know which media server (preferably OpenSource) to use? Is SEMS the only way to write an application on top of SER or there are other ways? Also, does anybody know how the applications in SEMS perform can they be deployed as a carrier grade solution? Regards --Srinivas Kotamarti

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Re: [OpenSER-Users] OpenSER as SIP Proxy for PSTN Gateway
by Andy Smith 13 Dec '07

13 Dec '07

>Register is used JUST to **RECEIVE** calls, not to send calls. > >SIP uses http-digest autentication. This means: in **EVERY** INVITE a proxy >wnat to authenticate the client (Asterisk) receives a challenge from the >proxy and the client must generate a hash (a response) with its credentials >and the nonce received.Ok, thanks for the reply Iñaki. So in the case of a configuration like mine what are other people doing to negate theneed to define all the users in both Asterisk and OpenSER? Are you authenticating the users on OpenSER basedon source IP? Or allowing all connections without authentication? And in general any pointers on the best way for me to achieve a simple accountig solution that is easy to maintain as more asterisk boxes and users are added?

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Re: [Serusers] Can SER be used as a SIP A/S
by Jiri Kuthan 13 Dec '07

13 Dec '07

At 19:54 13/12/2007, Srinivas Kotamarti wrote: >Content-Language: en-US >Content-Type: multipart/alternative; > boundary="_000_95BADEF3CDEB8441838C6AD39FA619A31343E206C0NAMAEMAILnmss_" > >All, >Can SER be used as a SIP A/S for example to write a carrier grade Ringback Tone application on top of SER. > >Does anybody know which media server (preferably OpenSource) to use? > >Is SEMS the only way to write an application on top of SER or there are other ways? Use SEMS. > >Also, does anybody know how the applications in SEMS perform can they be deployed as a carrier grade solution? Very well. -jiri > > >Regards > >--Srinivas Kotamarti > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] Openser + radiusclient accounting need more information
by Fernando Yubero 13 Dec '07

13 Dec '07

Hi group, Im new in the mailing list, and starting again with Openser, last time I used was first release like an alternative to SER. Later dont run any SIP service. Actually Im testing OpenSer for a new service, voice service with PSTN, etc, and Im testing accounting via radius. I use this help to work basic things. www.openser.org/docs/openser-radius-1.0.x.html thanks for this documentation very helpful for me. But I dont get same result. My version Openser 1.2.2. I get this fields in start and stop, very less than example. Wed Dec 12 17:33:12 2007 Acct-Status-Type = Start Service-Type = SIP Sip-Response-Code = 200 Sip-Method = 1 Event-Timestamp = "Dec 12 2007 17:33:05 ART" Sip-From-Tag = "94ab8960-c82d7718-13c4-7db17-4cc178eb-7db17" Sip-To-Tag = "94b908a0-c82d7717-13c4-3815-ecb28f3-3815" Acct-Session-Id = "94ab6a20-c82d7718-13c4-7db17-7fcaa8b-7db17(a)elmejorcable.com.ar" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xxx.xxx.xxx.3 Client-IP-Address = xxx.xxx.xxx.3 Acct-Unique-Session-Id = "49ff3575432365d6" Timestamp = 1197491592 Wed Dec 12 17:33:19 2007 Acct-Status-Type = Stop Service-Type = SIP Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = "Dec 12 2007 17:33:11 ART" Sip-From-Tag = "94b908a0-c82d7717-13c4-3815-ecb28f3-3815" Sip-To-Tag = "94ab8960-c82d7718-13c4-7db17-4cc178eb-7db17" Acct-Session-Id = "94ab6a20-c82d7718-13c4-7db17-7fcaa8b-7db17(a)elmejorcable.com.ar" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xxx.xxx.xxx.3 Client-IP-Address = xxx.xxx.xxx.3 Acct-Unique-Session-Id = "49ff3575432365d6" Timestamp = 1197491599 I think is about default fields to send for accounting via radius, How to get full info??? With radius_extra modparam??? Thanks in advanced. Fernando Yubero Communication Partner

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[OpenSER-Users] Preparing the release of OpenSER 1.3 "BlackCat"
by Bogdan-Andrei Iancu 13 Dec '07

13 Dec '07

Hi, As scheduled, tomorrow will be the official release of OpenSER version 1.3. NAME ===== Following the proposal and the feedback from the lists, we will start to label each major version number with a name - it is more friendly . Note that the version numbering will still be the primary ID. OpenSER 1.3 will be named "BlackCat". CODE ===== As far as I checked the lists and tracker, all the critical and important bugs were fixed. No pending issues, as far as I know. Tomorrow I will run some compiling tests on several arch and OS to see if any warning is reported. PS: does anybody offer some access on SUN+SOLARIS ? As there are only a small number of pending changes on SVN (warnings and docs), probably in the afternoon, the 1.3 branch will be created. DOCS ===== The module documentation is up to date, thanks to the policy of synchronization between the code and doc updates. Just in case, I will re-generate all the text READMEs from the SGML files. Thanks to Henning, the changelog and Credits files were updated on SVN. Thanks to Daniel, we have a start for the the "New Features" document (see http://www.openser.org/dokuwiki/doku.php/sandbox:new-in-1.3.x). The idea will be to have it finalized y tomorrow and imported in the web pages (openser.org and SF). There will be a need for a overview of the "INSTALL" helper in order to be updated for 1.3 (maybe the DB part only). I suggest to put together all the docs from openser and upload them also on SF - of course this is not a priority, it can be done later, in the following days. TARBALLS ======== Tarballs with sources and binaries (linux i386) will be uploaded on openser.org and SF sites. Of course after the creation of the new 1.3 branch on SVN. Any other contributions (binaries of other OSs or archs) are welcomed - just send an email to team(a)openser.org . PACKAGES ======== After the official release, specs will be updated (if not yet) and packages will be generated. Please help with this process and let us know if you can generate any packages. Again use the lists or team(a)openser.org emails. I will generate the debian packages (unstable, stable, testing). The stable package will be also pushed on the openser.org APT repo. SITES UPDATES =========== Both openser.org and SF sites need to be updated for with news, docs, downloads, etc. This probably will take couple of days. Please let me know if there is something missing in the above list. Otherwise, if people are able and willing with some of the tasks, again, just let me know. Otherwise, have the champaign on cold for tomorrow ;)... Regards, Bogdan

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Re: [OpenSER-Users] Could someone help me with WeSIP ?
by Daniel Grotti 13 Dec '07

13 Dec '07

Hi Elias, This is the WeSIP log's file, when I run startup.sh: 2007-12-12 10:02:57 HttpConnector Opening server socket on host IP address PROXY_IP 2007-12-12 10:02:57 Starting service Tomcat-Standalone 2007-12-12 10:02:57 HttpConnector[8080] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][0] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][1] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][2] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][3] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][4] Starting background thread 2007-12-12 10:02:57 ConvergedEngine[Standalone]: Converged Application Server SIP&HTTP-0.1 2007-12-12 10:02:57 HostConfig[PROXY_IP]: Deploying configuration descriptor manager.xml 2007-12-12 10:02:57 WebappLoader[/manager]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/manager 2007-12-12 10:02:57 WebappLoader[/manager]: Deploy JAR /WEB-INF/lib/dwr.jar to /usr/src/wesip/wesipapps/../server/webapps/manager/WEB-INF/lib/dwr.jar 2007-12-12 10:02:57 WebappLoader[/manager]: Deploy JAR /WEB-INF/lib/jrobin-1.4.0.jar to /usr/src/wesip/wesipapps/../server/webapps/manager/WEB-INF/lib/jrobin-1.4.0.jar 2007-12-12 10:02:57 WebappLoader[/manager]: Deploy JAR /WEB-INF/lib/statistics.jar to /usr/src/wesip/wesipapps/../server/webapps/manager/WEB-INF/lib/statistics.jar 2007-12-12 10:02:58 ContextConfig[/manager]: Missing application sip.xml, using defaults only 2007-12-12 10:02:58 ContextConfig[/manager]: Configured an authenticator for method BASIC 2007-12-12 10:02:58 StandardManager[/manager]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:58 StandardManager[/manager]: Seeding of random number generator has been completed 2007-12-12 10:02:58 StandardHttpWrapper[/manager:default]: Loading container servlet default 2007-12-12 10:02:58 default: init 2007-12-12 10:02:58 dwr-invoker: init 2007-12-12 10:02:58 StandardHttpWrapper[/manager:invoker]: Loading container servlet invoker 2007-12-12 10:02:58 invoker: init 2007-12-12 10:02:58 jsp: init 2007-12-12 10:02:58 statistics: init 2007-12-12 10:02:58 StandardHttpContext[/manager]: StandardHttpContext [WeSIP Manager Application] has been started 2007-12-12 10:02:58 HostConfig[PROXY_IP]: Expanding web application archive click2call.sar 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding of random number generator has been completed 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding of random number generator has been completed 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/click2call 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy class files /WEB-INF/classes to /usr/src/wesip/wesipapps/click2call/WEB-INF/classes 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/activation.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/activation.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/axis-ant.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/axis-ant.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/axis.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/axis.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/commons-discovery-0.2.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/commons-discovery-0.2.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/commons-logging.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/commons-logging.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/dwr.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/dwr.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/jaxrpc.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/jaxrpc.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/log4j-1.2.8.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/log4j-1.2.8.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/mail.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/mail.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/saaj.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/saaj.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/wsdl4j-1.5.1.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/wsdl4j-1.5.1.jar 2007-12-12 10:02:59 StandardManager[/click2call]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:59 StandardManager[/click2call]: Seeding of random number generator has been completed 2007-12-12 10:02:59 StandardHttpWrapper[/click2call:default]: Loading container servlet default 2007-12-12 10:02:59 default: init 2007-12-12 10:02:59 init 2007-12-12 10:02:59 dwr-invoker: init 2007-12-12 10:02:59 init 2007-12-12 10:02:59 AxisServlet: init 2007-12-12 10:03:00 StandardHttpWrapper[/click2call:invoker]: Loading container servlet invoker 2007-12-12 10:03:00 invoker: init 2007-12-12 10:03:00 jsp: init 2007-12-12 10:03:00 AdminServlet: init 2007-12-12 10:03:00 SOAPMonitorService: init 2007-12-12 10:03:00 ConvergedContext[/click2call]: ConvergedContext [Click2Call SipServlet] has been started 2007-12-12 10:03:00 HostConfig[PROXY_IP]: Deploying web application directory click2call-08_11_06 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/click2call-08_11_06 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy class files /WEB-INF/classes to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/classes 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/activation.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/activation.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/axis-ant.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/axis-ant.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/axis.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/axis.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/commons-discovery-0.2.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/commons-discovery-0.2.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/commons-logging.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/commons-logging.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/dwr.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/dwr.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/jaxrpc.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/jaxrpc.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/log4j-1.2.8.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/log4j-1.2.8.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/mail.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/mail.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/saaj.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/saaj.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/wsdl4j-1.5.1.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/wsdl4j-1.5.1.jar 2007-12-12 10:03:01 StandardManager[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:01 StandardManager[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-12-12 10:03:01 StandardHttpWrapper[/click2call-08_11_06:default]: Loading container servlet default 2007-12-12 10:03:01 default: init 2007-12-12 10:03:01 init 2007-12-12 10:03:01 dwr-invoker: init 2007-12-12 10:03:01 init 2007-12-12 10:03:01 AxisServlet: init 2007-12-12 10:03:01 StandardHttpWrapper[/click2call-08_11_06:invoker]: Loading container servlet invoker 2007-12-12 10:03:01 invoker: init 2007-12-12 10:03:01 jsp: init 2007-12-12 10:03:01 AdminServlet: init 2007-12-12 10:03:02 SOAPMonitorService: init 2007-12-12 10:03:02 ConvergedContext[/click2call-08_11_06]: ConvergedContext [Click2Call SipServlet] has been started 2007-12-12 10:03:02 HostConfig[PROXY_IP]: Deploying web application directory ROOT 2007-12-12 10:03:02 ManagerBase[]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:02 ManagerBase[]: Seeding of random number generator has been completed 2007-12-12 10:03:02 ManagerBase[]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:02 ManagerBase[]: Seeding of random number generator has been completed 2007-12-12 10:03:02 WebappLoader[]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/_ 2007-12-12 10:03:02 ContextConfig[]: Missing application sip.xml, using defaults only 2007-12-12 10:03:02 StandardManager[]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:02 StandardManager[]: Seeding of random number generator has been completed 2007-12-12 10:03:02 StandardHttpWrapper[:default]: Loading container servlet default 2007-12-12 10:03:02 default: init 2007-12-12 10:03:02 StandardHttpWrapper[:invoker]: Loading container servlet invoker 2007-12-12 10:03:02 invoker: init 2007-12-12 10:03:02 jsp: init 2007-12-12 10:03:02 ConvergedContext[]: ConvergedContext [Welcome] has been started 2007-12-12 10:03:02 > Default Wesip Application ==> http://PROXY_IP:8080/ > 2007-12-12 10:03:02 > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml I'd like to use " Click to dial " service. I don't write configuration file yet (sip.xml or web.xml). At this point, What should I do? Shouldn't I to access to Manager Wesip Application? (Manager Wesip Application ==> http://PROXY_IP:8080/manager/html?) - If yes, in this case I get an 500 http Error. My problem is very simple : I'm new in using servlet so I 'd like to know how I have to move (I didn't find no useful guide on this) . thanks. daniel -----Messaggio originale----- Da: elias baixas [mailto:elias.baixas@voztele.com] Inviato: mer 12/12/2007 0.40 A: Daniel Grotti Cc: users(a)lists.openser.org Oggetto: Re: [OpenSER-Users] Could someone help me with WeSIP ? Hello Daniel, can you provide some WeSIP's logs and openser's logs to gain some more information on the problem ? it's hard to guess what might be wrong with just the configuration. of course, you can substitute your IP addresses with some predefined token for the sake of confidentiality. for example: bash#>sed -i -e "s/192.168.1.1/PRIVATE_PROXY_IP/g" openser.cfg thanks !

3 2

[OpenSER-Users] Any toturial or tip for using Seas module and wesip Ringtone application
by CHEN XUEQIN 13 Dec '07

13 Dec '07

Hi: I just installed wesip application server and 1.3.0-pre1-notls with seas.so modules. I also downloaded ringtone-24_05_07.sar application into path $WESIP_ROOT/wesipapps. Ijust configed openser.cfg and server.xml especially for those key property such as com.voztele.javax.sip.SER_ADDRESS, listen_sockets. The openser.cfg was mainly copied from http://www.openser.org/docs/modules/1.3.x standalone script. It seemed that openser and wesip server started correctly. Then I have the following test: 1. UA1 register to openser 2. UA2 register to openser 3. UA1 call UA2 through openser 4. UA1 ringing, received 100 trying, no 183 with SDP Can anyone tell me how to use ringtone applicaton? Any config should be done with ringtone application? Or my setup was wrong? Thanks in advance. Best Regards, Chen Xueqin

1 0

[OpenSER-Users] ORACLE-PORT
by Martin.Gruber1＠infineon.com 13 Dec '07

13 Dec '07

HI @ all, I have a short question: is it possible to add the Port to the Connectionstring of unixODBC in openser.cfg ? Thanks! Martin

2 1

[OpenSER-Users] migrating 1.2.x to 1.3.0
by Daniel-Constantin Mierla 13 Dec '07

13 Dec '07

Hello, those that experienced upgrades from 1.2.x to upcoming 1.3.0, may want to contribute to next wiki page, with found issues and their solutions: http://www.openser.org/dokuwiki/doku.php/install:1.2.2-to-1.3.0 Cheers, Daniel

1 0

[Serusers] acc module is not working wiht mysql module
by Rogelio Serrano 13 Dec '07

13 Dec '07

I think this is a bug. If the the accounting messages show up in syslog then it should also show up in ser.acc. -- Lay low and nourish in obscurity

1 0

[Serusers] mysql acc table empty
by Rogelio Serrano 13 Dec '07

13 Dec '07

I am totally stumped with the accounting module not inserting any data into the mysql "acc" table. I can see the accounting entries in syslog. There are no errors with mysql. I have started reading the source code as a last resort but i dont have enough time left. here is my config: debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) listen=sip.asianet.co.in /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 60) # Ping only contacts that are known to be # behind NAT modparam("nathelper", "ping_nated_only", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # special handling for NATed clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence of received if (nat_uac_test("3")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP log("LOG: fixing nated sdp"); }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); #route(1); t_relay(); break; }; if (method=="INVITE"){ setflag(1); } if (method=="BYE"){ setflag(1); } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; #if (uri=~"^sip:1[0-9]*@") { # International PSTN # route(2); # break; #}; # native SIP destinations are handled using our USRLOC DB if (uri=~"^sip:8[0-9]*@") { # ip to ip if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); route(1); break; } else { sl_send_reply("404", "Not Found"); break; }; } else { #send to pstn gateway route(2); }; }; } route[1] { # # Forcing media relay if necessary if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { force_rtp_proxy(); t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); }; # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- setflag(1); rewritehostport("192.168.1.1:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; } -- Lay low and nourish in obscurity

1 0

[OpenSER-Users] WeSIP: Not SER connector available - Check the Configuration
by Franz Edler 13 Dec '07

13 Dec '07

Hi, I try to configure WeSIP which should not be so hard according to documentation. When I start OpenSER and WeSIP I get on the logs of OpenSER: -1(0) Fantastic, we have a new client: wesip1 -1(0) Fantastic, we have a new client: wesip1 -1(0) EUREKA, we have a new completed AS: wesip1 Also on WeSIP log I get: ExpressConnector [main] - SEAS module successfully connected at 192.168.0.5:5080 But when I call the manager Web-GUI I get as a status of OpenSER: Not SER connector available - Check the Configuration This seems to be in contradiction to above logs, isn't it? Any comment appreciated Franz

1 0

[OpenSER-Users] OpenSER as SIP Proxy for PSTN Gateway
by Andy Smith 12 Dec '07

12 Dec '07

Hi, As you may work out Im not an expert in this, hopefully for those of you that are this is a pretty straight forward question. Im trying to implement something like this: Cisco_as5300 <-----> OpenSER <------> Atersisk PSTN Gateway SIP Proxy PBX ^ Radius I am attempting to setup OpenSER as a SIP proxy for a Cisco PSTN Gateway, on the other side of OpenSER I have an Asterisk system. My confusion centres around the connection between the end users (SNOM or whatever) which hang off Asterisk and the connection between Asterisk and OpenSER. I have configure Asterisk to SIP register to OpenSER, all works ok here and Asterisk sees OpenSER as a SIP peer and channel. The bit Im not getting is that when someone dials a number from a handset connected to Asterisk that needs to route out via OpenSER, Asterisk seems to pass the username and password credentials of the handset to OpenSER. Firstly if its going to do this, then what is the point of SIP registering Asterisk as its authenticating ever session as individual entities anyway, secondly this isnt scaleable :( it seems to mean I have to configure each SNOM user both in Asterisk and in OpenSER. All of which doesnt seem very logical, can anyone tell me if I'm doing something wrong or if Im approaching this in the wrong way or if I need to approach the problem in some other way... The end goal of this is that Id like the ability to bill on a per SIP peer basis customer Asterisk boxes, I dont want to end up in a situation were I have to know about every SNOM a customer has. many many thanks in advance, Im a bit stuck at the mo! thanks Andy.

2 1

[Serusers] SER Incoming Calls
by Arun Kumar 12 Dec '07

12 Dec '07

HI User ----> Broadsoft -----> SER (Added Broadsoft ip as Trusted ) ------> Ser exten (Behind NAT) from SER exten I can dial my Broadsoft exten works fine. But reverse is not working fine. Incoming call from Broadsoft exten to my ser exten (027113) if (uri=~"^sip:027113@") { if (lookup("location")) { route(4); route(1); break; }; };

1 0

[OpenSER-Users] openser 1.3 packaging
by Daniel-Constantin Mierla 12 Dec '07

12 Dec '07

Hello, for those that want to start updating the spec files for packages and test them, I have created a README file that lists the new modules and their dependencies, as well as suggestions for where to place the modules. Also, I have uploaded today's source tarball, so all elements are in single place. Please check: http://www.openser.org/pub/openser/pre-1.3.0/ If you need further help, let me know. Comments and suggestions are welcome. Cheers, Daniel

4 3

[Serusers] SER Strange Calling Issue
by Arun Kumar 12 Dec '07

12 Dec '07

Hi User ---> SER -----> BroadoSoft -----> PSTN when I send calls to PSTN I'm changing the From to one of BroadSoft users b'coz without that BroadSoft don't allow me to Call PSTN. When call comes to PSTN Route I'm changing the the From : subst('/^From:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/From:< sip:8233843435@Broadsoft IP>/ig'); calls goes fine to Broadsoft and then to PSTN but, once call connect to PSTN I'm able to hear other side but my phone always shows Calling and other side is not able to hear me (b'coz from my side channel is not getting answered). I tried forcing mediaproxy before PSTN Calls but no luck. Please advise. Its very urgent. Thanks, Arun

1 1

[OpenSER-Users] Openser 1.3 howto
by Fabio Isgro 12 Dec '07

12 Dec '07

How to install openser 1.3 there is any howto out there? Best Regards -- -\\_ Fabio Isgro' -}}\_ Babel S.r.l. - http://www.babel.it -}}}\-__ Tel. +39.06.91801075 }\/ -}}}\\°_\ Fax +39.06.91612446 }/\ }}}}\\ _/ Voip sip:9093@voip.babel.it ------ P.zza S.Benedetto da Norcia, 33 - 00040 Pomezia (Roma) B ° a ° b° e° l "CONFIDENZIALE: Questo messaggio ed i suoi allegati sono di carattere confidenziale per i destinatari in indirizzo. Se hai ricevuto questo messaggio per errore sei invitato cortesemente a rispondere immediatamente al mittente e cancellare tutti i suoi contenuti."

2 1

[OpenSER-Users] CPL DTD
by Daniel Grotti 12 Dec '07

12 Dec '07

Hi Bogdan, Cpl modules is based on DTD file. It doesn't support namespaces. So how could I make extensions (for example add a new node) on the dtd file remaining RFC3880 compliant ? Is it possible? Regards, daniel

1 0

[OpenSER-Users] Display Registered Users
by VoIP Forums www.Go4Calls.com 12 Dec '07

12 Dec '07

Hi All, Could you please guide us with right cmd to display the users which are registered successfully with openser? Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 2

[OpenSER-Users] Question about avp_aliases and script_flags
by Helmut Kuper 12 Dec '07

12 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, avpops doc say that avp spec for integer values is $avp(i[script_flags]:<ID>) Is it possible to set avp aliases to that values, e.g. avp_aliases="x=i1:10" avp_aliases="y=i2:10" ? When I do that, I got this error message in openser 1.3: Nov 14 16:00:42 [28355] ERROR:core:add_avp_galias: duplicate alias/avp entry Nov 14 16:00:42 [28355] ERROR:core:add_avp_galias_str: add global alias failed Nov 14 16:00:42 [28355] CRITICAL:core:yyerror: parse error (54,3-4): invalid AVP aliases regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHOxLU4tZeNddg3dwRArlWAKCd7mGq6K3yODf/8bvZRa9F/vcpugCfWsb7 Ale6eO7Xz8C0fnvyg5m4/n0= =tPWq -----END PGP SIGNATURE-----

2 9

[OpenSER-Users] Could someone help me with WeSIP ?
by Daniel Grotti 11 Dec '07

11 Dec '07

Hi all, I'm trying to use WeSIP but I don't be able to connect to "Manager" link. I receive a 500 Http error. This is my server.xml file: where I hide Proxy IP Address with PROXY_IP and in OpenserSER I have setted modparam("seas", "listen_sockets","localhost:5080") ***************************************************************************************** <Server port="8005" shutdown="SHUTDOWN" debug="0"> <Service name="Tomcat-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" address="PROXY_IP" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="PROXY_IP:5060" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="localhost" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5080" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="javax.sip.STACK_NAME" value="app_server_one" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="PROXY_IP" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="PROXY_IP" nameSip="PROXY_IP_5060" debug="0" appBase="wesipapps" port="5060" className="com.voztele.sipservlet.core.ConvergedHost" unpackWARs="true"> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/>  </Host> </Engine> </Service> </Server> ************************************************************************************************************* Regards, daniel

2 1

[OpenSER-Users] Softphones supporting "pua_bla"?
by Iñaki Baz Castillo 11 Dec '07

11 Dec '07

Hi, I'd like to try the "pua_bla" module but don't have a SNOM or Polycom (models I think they support BLA). I've a Thomson S2030 but it does BLF (it doesn't send a REGISTER, just a SUBSCRIBE with "Event: dialog"). Is there any way to try the "pua_bla" module? any softphone supporting it? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

4 6

[OpenSER-Users] FW: No user and password are stored in the openxcap db, how do i do?
by KevinKinnan 11 Dec '07

11 Dec '07

1st question: There is no user and pwd in the db for the first time use. if I want to send a http request to openxcap what user name and password should i use? BTW, openims core and openser(presence module) are using in front of openxcap. I atteched a trace that still showed 401 after digesting request(by default, the username and password are both alice). 2nd question: I am designing a j2me programe to connect to openxcap, but only GET, POST, HEAD methods are supoorted in j2me. Can I use POST instead of PUT for request sending? THANK U! Regards, Kevin 使用新一代 Windows Live Messenger 轻松交流和共享！立即体验！ _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

Re: [Serusers] Loading attributes question
by Jiri Kuthan 11 Dec '07

11 Dec '07

I'm afraid there is no easy way for doing so -- the multiple-value AVPs have been seen as "sets", disregarding the order. I think one would have to have a new function like order_avp($avp)... At 00:05 10/12/2007, Steven C. Blair wrote: >Content-Language: en-US >Content-Type: multipart/alternative; > boundary="_000_EBC0E89A333CED46A6273F415B6386E814BA4E6EEBMAILBOXexchan_" > > > >I have a locally defined table with attributes which may contain several values. I define each attribute /value pair as a new entry in the table then use avp_load to retrieve these values from the database. The problem is that I don't know how to retrieve the values in a specific order. For example if I have a=1, a=2, a=3 when I load the values for "a" they may be 2,1,3 or 3,1,2, etc. Is there any way to force loading in a specific sequence. > >Thanks,Steve >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[OpenSER-Users] users@lists.openser.org
by minz1 11 Dec '07

11 Dec '07

Hi Schumann Sebastian, I too have the same problem regarding XCap Server project. ---------------------------------------------------------------------------------------- Following are the trace generated after running "python2.4 openxcap --no-fork" command:- ---------------------------------------------------------------------------------------- Starting Open XCAP 0.9.7 Traceback (most recent call last): File "/usr/local/bin/openxcap", line 56, in ? from xcap.server import XCAPServer File "/usr/local/lib/python2.4/site-packages/xcap/server.py", line 21, in ? from xcap import authentication File "/usr/local/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? from xcap.appusage import getApplicationForURI File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 491, in ? applications = {'xcap-caps': XCAPCapabilitiesApplication(), File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema ------------------------------------------------------------------------------------------ Following are the files which we are using:= ================================= openxcap-0.9.7 libxml2-2.6.30 twisted-web2-0.2.0+svn20070403.orig Python-2.4.2 lxml-1.3.4 zope.interface-3.3.0 python-application-1.0.9 MySQL-python-1.2.2 ================================= I followed the instructions from the openxcap webpage and did what instructed there but not able to make it work. Please help me. I will be always greatful to that help. regds Suraj Schumann Sebastian wrote: Dear all I have a big problem with openxcap. I don't get it work :( As I saw on their webpage, this mailing list can be posted for this concern as well. I am using version 0.9.3 and acc. the installation file on http://www.openxcap.org all current dependencies from the Debian unstable distribution. When I start /usr/bin/openxcap, the following error occurs in the syslog: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema As I wrote, I followed all instructions from the openxcap webpage but didn't find any hints how to make it work. Any help on that would be appreciated. Best regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) -- View this message in context: http://www.nabble.com/users%40lists.openser.org-tp14279161p14279161.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] users@lists.openser.org
by minz1 11 Dec '07

11 Dec '07

Hi Schumann Sebastian, I too have the same problem regarding XCap Server project. ---------------------------------------------------------------------------------------- Following are the trace generated after running "python2.4 openxcap --no-fork" command:- ---------------------------------------------------------------------------------------- Starting Open XCAP 0.9.7 Traceback (most recent call last): File "/usr/local/bin/openxcap", line 56, in ? from xcap.server import XCAPServer File "/usr/local/lib/python2.4/site-packages/xcap/server.py", line 21, in ? from xcap import authentication File "/usr/local/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? from xcap.appusage import getApplicationForURI File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 491, in ? applications = {'xcap-caps': XCAPCapabilitiesApplication(), File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema ------------------------------------------------------------------------------------------ Following are the files which we are using:= ================================= openxcap-0.9.7 libxml2-2.6.30 twisted-web2-0.2.0+svn20070403.orig Python-2.4.2 lxml-1.3.4 zope.interface-3.3.0 python-application-1.0.9 MySQL-python-1.2.2 ================================= I followed the instructions from the openxcap webpage and did what instructed there but not able to make it work. Please help me. I will be always greatful to that help. regds Suraj Schumann Sebastian wrote: Dear all I have a big problem with openxcap. I don't get it work :( As I saw on their webpage, this mailing list can be posted for this concern as well. I am using version 0.9.3 and acc. the installation file on http://www.openxcap.org all current dependencies from the Debian unstable distribution. When I start /usr/bin/openxcap, the following error occurs in the syslog: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema As I wrote, I followed all instructions from the openxcap webpage but didn't find any hints how to make it work. Any help on that would be appreciated. Best regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) -- View this message in context: http://www.nabble.com/users%40lists.openser.org-tp14279160p14279160.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Serusers] rewritehostport and RFC3261 compliant
by inge 11 Dec '07

11 Dec '07

Hi all, I find an issue with a problem encountered on some products. The problem only concerns on-net calls. In fact, when a SER User A calls a SER User B : - if B doesn't answer, after the tm timer : ==> t_on_failure("1") ==> setflagt(24) ==> load $ruri/username in vm_uri/usr_vm if flag is set to 24 then ==> avp_pushto $ruri s:vm_uri ==> resetflag(24) ==> add prefix Prefix("710") ==> rewritehostport(Asterisk:4060) ==> append_branch ==> t_relay_to_udp(Asterisk,4060) This is the process apply for transfer the call to the voicemail Asterisk. But, and only in this on-net case, the To-Tag change during this process, cause the failure of communication for some products. Whereas the RFC 3261 specify "... This tag will be incorporated by both endpoints into the dialog and will be included in all future requests and responses in this call ...". Thanks for your support.

3 6

[OpenSER-Users] new stuff in 1.3.0
by Daniel-Constantin Mierla 11 Dec '07

11 Dec '07

Hello, I started a dokuwiki page to collect the new things in openser 1.3.x, I still have a lot to add, however, since there are new modules and enhancements from many developers, it is better that everybody contribute to this page, to have a good image of what is new in the upcoming release. I am sure I will be forgetting some. The page will be used in building release announcement. http://www.openser.org/dokuwiki/doku.php/sandbox:new-in-1.3.x Thanks, Daniel

1 0

[Serusers] media proxy manage NAT IP Change?
by Gonzalo J. Sambucaro 11 Dec '07

11 Dec '07

Hi, I would like to know if the Media Proxy Server is able to manage the NAT IP change of an established rtp stream and continue working with the new NAT IP without problems? Saludos, Gonzalo. -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4808585 MSLC gonzalo(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

1 0

[OpenSER-Users] trunk - openserro password not working
by Richard Bennett 11 Dec '07

11 Dec '07

Hi, After moving a working openser config for trunk to a new server, and updating the sources, I kept getting the error: ERROR:mysql:db_mysql_new_connection: driver error: Access denied for user 'openserro'@'localhost' (using password: YES) I double-checked, and the settings in: /usr/local/etc/openser/openserctlrc were set correct: DBROPW=newpassword So I set the mysql password for openserro back to the default 'openserro' in mysql, and it works, even with a different value in openserctlrc. Is this a known issue, bug, or am I doing something wrong? Richard.

2 3

[OpenSER-Users] Set CallerID as Username
by VoIP Forums www.Go4Calls.com 11 Dec '07

11 Dec '07

Hi All, i was routing calls from openser to asterisk to make accounting in asterisk side. I need small help, how i can set callerid = registrar username so in asterisk server i can capture the callerid to do the billing etc.... Thank You _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

2 1

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 11 Dec '07

11 Dec '07

Hello, seems that you haven't updated the database structure. In 1.2, there was a refurbishment that included primary keys, a.s.o. As I could guess from patch, you still use the old database schema. Cheers, Daniel PS. Please CC to mailing list, the answer can come much faster, with more competence from people experiencing same issues, and will help new comers with same case. On 12/10/07 15:07, Mihail Gluhowchenko wrote: > Hello; > Sorry %) > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX > 2007/12/10, Daniel-Constantin Mierla <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>>: > > Hello, > > can you please send unified format for the patch? > > diff -u oldfile newfile > > Thanks, > Daniel > > > On 12/10/07 14:29, Mihail Gluhowchenko wrote: > > Hello; > > Here it is sc.patch; > > my version is 'openser-1.2.2-notls' > > -- > > Regards, > > Mihail Gluhowchenko. > > Enterprise ELTEX. > > > > 2007/12/10, Daniel-Constantin Mierla < daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro> > > <mailto:daniel@voice-system.ro <mailto:daniel@voice-system.ro>>>: > > > > > > > > On 12/10/07 07:04, Mihail Gluhowchenko wrote: > > > Hello all work; :) > > was your version old or there is a bug in sc.dbtext? Can you > post the > > patch of changes? > > > > Best regards, > > Daniel > > > > > > > in script sc.dbtext use old format subscriber. > > > Little changes and all work > > > > > > date_now=$(date +"%s") > > > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | > > sed -e > > > 's/ //') > > > > > > add) > > > if [ $# -ne 4 ] ; then > > > usage > > > exit 1 > > > fi > > > shift > > > credentials $1 $2 > > > is_user $1 > > > if [ $? -eq 0 ] ; then > > > echo -e "\nuser already exists\n" > > > exit 1 > > > fi > > > #Add to SUBSCRIBER file > > > #id(int,auto) username(str) domain(str) > > password(str) > > > first_name(str,null) last_name(str,null) email_address(str) > > > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > > > rpid(str,null) > > > > > > > > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > > > > > > echo $line >> $SUB_FILE > > > #Add to URI file > > > #id(int,auto) domain(str) uri_user(str) > > > last_modified(int) > > > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > > > echo $line >> $URI_FILE > > > echo -e "\nNew user ($SERUSER@$SERDOMAIN) > added to > > > DBTEXT files\n" > > > ;; > > > -- > > > Regards, > > > Mihail Gluhowchenko. > > > Enterprise ELTEX. > > > > > > > > >

2 1

[OpenSER-Users] Loop Detection
by Daniel Corbe 11 Dec '07

11 Dec '07

I have a problem. I'm taking INVITES from a PortaSIP proxy and handing it back with a new request URI and To: header; however, their loop detection algorithm is calling this a loop. Is there a way to fix this from within OpenSER? I'm assuming my proxy would have to send an INVITE with a completely new branch tag and From tag? 10 Dec 14:52:30.831/GLOBAL/b2bua: SENDING message to 206.71.169.114:5060: INVITE sip:*94310@206.71.169.114:5060 SIP/2.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport Max-Forwards: 70 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:*94310@206.71.169.114> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 h323-conf-id: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.841/GLOBAL/b2bua: RECEIVED message from 206.71.169.114:5060: SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:*94310@206.71.169.114> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Server: OpenSER (1.2.2-notls (i386/freebsd)) Content-Length: 0 10 Dec 14:52:30.842/GLOBAL/ser[69804]: RECEIVED message from 206.71.169.114:5060: INVITE sip:1a10000000009@216.151.143.69:5060 SIP/2.0 Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Max-Forwards: 69 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 h323-conf-id: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp P-hint: rr-enforced v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.842/722814-3406315950-714741@msx69.mydomain.com~1o/ser[69804]: processing INVITE received from 206.71.169.114 10 Dec 14:52:30.843/GLOBAL/ser[69804]: SENDING message to 206.71.169.114:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 10 Dec 14:52:30.844/GLOBAL/ser[69804]: SENDING message to 216.151.143.69:5061: INVITE sip:1a10000000009@216.151.143.69:5061 SIP/2.0 Record-Route: <sip:216.151.143.69;ftag=3b383c39c2bc93d8106f4425074ef399;lr> Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> Via: SIP/2.0/UDP 216.151.143.69;branch= z9hG4bK961c.003d31266db19cfeb34aa550d186a976.0 Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Max-Forwards: 16 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp P-hint: rr-enforced v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.844/GLOBAL/b2bua: RECEIVED message from 216.151.143.69:5060: INVITE sip:1a10000000009@216.151.143.69:5061 SIP/2.0 Record-Route: <sip:216.151.143.69;ftag=3b383c39c2bc93d8106f4425074ef399;lr> Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> Via: SIP/2.0/UDP 216.151.143.69;branch= z9hG4bK961c.003d31266db19cfeb34aa550d186a976.0 Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Max-Forwards: 16 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp P-hint: rr-enforced v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.847/GLOBAL/b2bua: SENDING message to 216.151.143.69:5060: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 216.151.143.69;branch= z9hG4bK961c.003d31266db19cfeb34aa550d186a976.0 Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Record-Route: <sip:216.151.143.69;ftag=3b383c39c2bc93d8106f4425074ef399;lr> Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Server: Sippy

2 5

Re: [OpenSER-Users] Problem with Mediaproxy
by Daniel 10 Dec '07

10 Dec '07

Hi, this is my call flow (from client side making the call: 192.168.10.224): The INVITE received by 192.168.0.167 (sent by 192.168.10.224 and processed by proxy) has SDP body with PRIVATE IP and PORT. ----------------------------------------------------------------------------------- # 192.162.10.224 ----> PROXY Request-Line: INVITE sip:franco.callegati@PROXY_IP SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.224:64244 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543- Max-Forwards: 70 Contact: <sip:daniel.grotti@192.168.10.224:64244> To: "franco.callegati"<sip:franco.callegati@PROXY_IP> From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1000",realm="PROXY_IP",nonce="47593c9fa32034a66fdb7493ff499c7215d4d8db",uri="sip: franco.callegati@PROXY _IP",response="82d21af18029aa02388869088c3619d2",algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 374 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 8 2 IN IP4 192.168.10.224 Session Name (s): CounterPath X-Lite 3.0 Connection Information (c): IN IP4 192.168.10.224 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 53624 RTP/AVP 107 119 100 106 0 105 98 8 101 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtpmap:107 BV32/16000 Media Attribute (a): rtpmap:119 BV32-FEC/16000 Media Attribute (a): rtpmap:100 SPEEX/16000 Media Attribute (a): rtpmap:106 SPEEX-FEC/16000 Media Attribute (a): rtpmap:105 SPEEX-FEC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): sendrecv # PROXY -----> 192.162.10.224 Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.10.224:64244 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543-;rport=28943;received= 85.18.14.25 To: "franco.callegati"<sip:franco.callegati@PROXY_IP> From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE Server: OpenSER (1.3.0-pre1-notls (i386/linux)) Content-Length: 0 # PROXY -----> 192.162.10.224 Status-Line: SIP/2.0 180 Ringing Message Header Via: SIP/2.0/UDP 192.168.10.224:64244;received=85.18.14.25 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543-;rport=28943 Record-Route: <sip:PROXY_IP:5060;lr;nat=yes;ftag=4554424c> Contact: <sip:franco.callegati@85.18.14.25 :42214;rinstance=96888d44783df4a0> To: "franco.callegati"<sip:franco.callegati@PROXY_IP>;tag=d06d8b2d From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 # PROXY -----> 192.162.10.224 Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.224:64244;received=85.18.14.25 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543-;rport=28943 Record-Route: <sip:PROXY_IP:5060;lr;nat=yes;ftag=4554424c> Contact: <sip:franco.callegati@85.18.14.25 :42214;rinstance=96888d44783df4a0> To: "franco.callegati"<sip:franco.callegati@PROXY_IP>;tag=d06d8b2d From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 372 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 7 2 IN IP4 192.168.0.167 Session Name (s): CounterPath X-Lite 3.0 Connection Information (c): IN IP4 192.168.0.167 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 45566 RTP/AVP 107 119 100 106 0 105 98 8 101 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtpmap:107 BV32/16000 Media Attribute (a): rtpmap:119 BV32-FEC/16000 Media Attribute (a): rtpmap:100 SPEEX/16000 Media Attribute (a): rtpmap:106 SPEEX-FEC/16000 Media Attribute (a): rtpmap:105 SPEEX-FEC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): sendrecv 192.162.10.224 ----> PROXY Request-Line: ACK sip:franco.callegati@85.18.14.25:42214;rinstance=96888d44783df4a0 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.224:64244 ;branch=z9hG4bK-d87543-66641951d8357408-1--d87543- Max-Forwards: 70 Route: <sip:PROXY_IP:5060;lr;nat=yes;ftag=4554424c> Contact: <sip:daniel.grotti@192.168.10.224:64244> To: "franco.callegati"<sip:franco.callegati@PROXY_IP>;tag=d06d8b2d From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 ACK Proxy-Authorization: Digest username="1000",realm="PROXY_IP",nonce="47593c9fa32034a66fdb7493ff499c7215d4d8db",uri="sip: franco.callegati@PROXY _IP",response="82d21af18029aa02388869088c3619d2",algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 RTP STREAM BETWEEN CLIENTS (MEDIAPROXY DOESN'T WORK) CALL ENDED ---------------------------------------------------------------------------------------------------------------------- this is my config file: debug=3 fork=yes log_stderror=yes listen=PROXY_IP port=5060 children=4 dns=no rev_dns=no disable_dns_blacklist=yes mpath="/usr/SER_proxy//lib/openser/modules/" #------------------------------------------------------ # Load Module #------------------------------------------------------ loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "cpl-c.so" loadmodule "alias_db.so" #---------------------------------------------------------- # Parameters #---------------------------------------------------------- modparam("auth_db|uri_db|usrloc|domain|alias_db", "db_url", " mysql://topenser:openserrw@localhost/topenser") modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("rr", "enable_full_lr", 1) modparam("alias_db", "use_domain", 1) # REGISTRAR MODULE modparam("registrar", "max_contacts", 2) # max number of contacts per URI modparam("registrar", "append_branches", 0) # if 0: R-URI will be overwritten with the highest-q rated contact and the rest will be left unprocessed. modparam("registrar", "default_q", 1000) # default contacts q-value set as 1 modparam("registrar", "received_avp","$avp(i:42)") # URI_DB MODULE modparam("uri_db", "use_uri_table", 1) # Specify if the "uri" table should be used for checkings instead of "subscriber" table # CPL MODULE modparam("cpl-c", "db_url", "mysql://topenser:openserrw@localhost/topenser") modparam("cpl-c", "cpl_table", "cpl") modparam("cpl-c", "cpl_dtd_file", "/usr/src/openser-1.2.2-notls /modules/cpl-c/cpl-06.dtd") modparam("cpl-c", "lookup_domain", "location") modparam("cpl-c","lookup_append_branches",1) modparam("cpl-c","proxy_route", 10) #NATHELPER MODULE modparam("nathelper", "rtpproxy_disable_tout", 120) modparam("nathelper", "rtpproxy_tout", 2) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", "$avp(i:42)") modparam("nathelper", "sipping_method", "OPTIONS") # Use OPTIONS every 10sec to keep NAT alive modparam("nathelper", "sipping_bflag", 6) modparam("nathelper", "sipping_from", "sip:PROXY_IP") #MEDIAPROXY MODULE modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/openser/rtp-asymmetric-clients") # TM MODULE modparam("tm", "fr_inv_timer", 120) #------------------------------------------------------------ #MAIN ROUTE #----------------------------------------------------------- route { xlog("L_DBG", "INFO: NEW REQUEST - METHOD=$rm R-URI=$ru FROM=$fu TO=$tu IP=$si ID=$ci\n"); #------------------------------------ # Sanity checks #------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too much Hops"); exit; }; if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; }; #------------------------------------ # RECORD-ROUTE #------------------------------------ if(method=="INVITE" && client_nat_test("7")) { record_route_preset ("PROXY_IP:5060;nat=yes"); xlog("L_DBG", "INFO: RECORD-ROUTE WITH NAT"); } else if (method!="REGISTER") { record_route(); }; #---------------------------------------- # CALL TEAR DOWN #---------------------------------------- if (method=="BYE") { end_media_session(); }; if (method=="CANCEL") { end_media_session(); }; #-------------------------------------------- # LOOSE ROUTE #-------------------------------------------- if (loose_route()) { xlog("L_DBG", "INFO: LOOSE ROUTE"); if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403","Unauthorized"); exit; }; consume_credentials(); if (client_nat_test("7") || search("^Route:.*nat=yes")) { setbflag(7); use_media_proxy(); }; }; route(1); exit; }; #------------------------------------------------- #CALL TYPE PROCESSING #------------------------------------------------- if (!uri==myself) { route(1); exit; }; if (uri==myself) { if (method=="REGISTER") { xlog("L_DBG", "INFO: REGISTER Message received"); route(2); exit; }; if (method=="ACK") { route(1); exit; }; if (method=="INVITE") { xlog("L_DBG", "INFO: INVITE Message received"); route(3); exit; }; if (method=="MESSAGE") { xlog("L_DBG", "INFO: MESSAGE method received"); route(4); exit; }; lookup("aliases"); if (!uri==myself) { route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; }; route(1); } #-------------------------------------------------------- # REGISTER PROCESSING #-------------------------------------------------------- route[2] { # Check credentials if(!www_authorize("","subscriber")) { www_challenge("","0"); exit; } else if(!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); # Check client natted - if yes, set flag 6 if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); xlog("L_DBG","INFO: Registration behind NAT"); xlog("L_DBG","INFO: Bflag 6, related to BRANCH=$T_branch_idx, is setted as $bf"); fix_nated_register(); force_rport(); }; if(!save("location")) { sl_reply_error(); xlog("L_DBG", "Registration of [$fU] Failed"); }; xlog("L_DBG", "INFO: Registration of [$fU] Complete."); xlog("L_DBG", "INFO: 200 OK has been sended"); } #---------------------------------------------------------- # INVITE PROCESSING #---------------------------------------------------------- route[3] { sl_send_reply("100", "Trying"); # Check if caller is natted, if yes set flag 7, rewrite Contact HF with Proxy address if(client_nat_test("7")) { setbflag(7); xlog("L_DBG", "INFO: CLIENT INVITE BEHIND NAT"); xlog("L_DBG","NOTICE: Bflag 7, related to BRANCH=$T_branch_idx of this INVITE, is setted as $bf"); if(force_rport()) { xlog("L_DBG","INFO: rport FORCED"); }; if(fix_nated_contact()) { xlog("L_DBG","INFO: Contact HF is now [$ct]"); }; # append_hf("P-hint: INVITE behind NAT \n"); }; # Check Credentials if(!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Unauthorized"); exit; }; lookup("aliases"); if(uri!=myself) { route(1); exit; }; consume_credentials(); # Check if r-URI is an alias. If yes, rewrite RURI with User's URI if(!alias_db_lookup("dbaliases")) { xlog("L_DBG", "The [$ru] is not an alias"); }; # Use Caller CPL "Outgoing" Script, if exist if(!cpl_run_script("outgoing", "force_stateful")) { sl_send_reply("500","Outgoing CPL script execution failed"); exit; }; # Find destination user and re-write R-URI with its Contact if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; xlog("L_DBG", "INFO: LOCATION HAS BEEN FOUND"); xlog("L_DBG","INFO: R-URI is now [$ru]"); t_on_reply("1"); # Use CPL "Incoming" script of the destination client if(!cpl_run_script("incoming","force_stateful")) { sl_send_reply("500", "Incoming CPL script execution failed"); exit; }; route(11); route(1); } # -------------------------------- # PROXY ROUTE - CPL SCRIPT LOADED # -------------------------------- route[10] { xlog("L_DBG", "INFO: ROUTE 10 - PROXY ROUTE \n"); # append_hf("P-hint: CPL ROUTE 10 - PROXY-ROUTE\n"); if (isbflagset(6) || isbflagset(7)) { xlog("L_DBG", "INFO: flag 6 or 7 setted \n"); if(!isbflagset(8)) { setbflag(8); }; if(use_media_proxy()) { xlog("L_DBG","NOTICE: MEDIAPROXY in use for the CALLER!!"); # append_hf("P-hint: Mediaproxy in USE for the Caller\n"); }; }; } # ----------------------------- # CONTROL FLAG - NO CPL SCRIPT # ----------------------------- route[11] { xlog("L_DBG", "INFO: ROUTE 11 - NO CPL SCRIPT"); # append_hf("P-hint: ROUTE 11 \n"); if(isbflagset(6) || isbflagset(7)) { xlog("L_DBG", "INFO: flag 6 or 7 setted"); if(!isbflagset(8)) { setbflag(8); }; if(use_media_proxy()) { xlog("L_DBG","NOTICE: mediaproxy in use for the caller!!"); # append_hf("P-hint: MEDIAPROXY in USE for the CALLER\n"); }; }; } # -------------------------------------------------- # MESSAGE PROCESSING # -------------------------------------------------- route[4] { if (uri!=myself) { sl_send_reply ("403", "Forbidden"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; route(1); } #---------------------------------------------------- # RELEY ROUTE #---------------------------------------------------- route[1] { if (!t_relay("0x01")) # don't send 100 trying { if (method=="INVITE" || method=="ACK") { end_media_session(); xlog("L_DBG","INFO: MEDIA SESSION ENDED"); }; sl_reply_error(); xlog("L_DBG", "INFO: SENT ERROR REPLY"); }; } #--------------------------------------------------------------------- # On Reply Route #--------------------------------------------------------------------- onreply_route[1] { xlog("L_DBG","INFO : ONREPLY ROUTE"); # append_hf("P-hint: ONREPLY ROUTE \n"); if (status=~"(180)|(183)|2[0-9][0-9]") { if (isbflagset(8) && !search("^Content-Lenght:[ ]*0")) { if(use_media_proxy()) { xlog("L_DBG","INFO: MEDIAPROXY in use for the CALLED!"); # append_hf("P-hint: Media Proxy in use for the Callee!\n"); }; }; }; if (client_nat_test("7") && status=~"(180)|(183)|(200)") { fix_nated_contact(); xlog("L_DBG","INFO: CONTACT HF has been CHANGED"); # append_hf("P-hint: CALLEE'S CONTACT HF has benn CHANGED\n"); }; } ---------------------------------------------------------------------------------------------------------------------- Regards, Daniel Hi Daniel, Can u post your openser configuration as well as a complete ngrep trace for the calls you have problems with? DanB -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 3

[Serusers] ser and rtpproxy bridging
by mashudi 10 Dec '07

10 Dec '07

Hi all, I'm going to setup ser 0.9.6 for bridging two vpn network 10.12.12.0/24 and 10.177.10.0/24 using rtpproxy and a pstn gateway 10.177.10.60 , if some one can have any sample configuration of some good guide to configure. Please help. best regards, mashudi

1 0

[OpenSER-Users] openxcap Problem
by Schumann Sebastian 10 Dec '07

10 Dec '07

Dear all I have a big problem with openxcap. I don't get it work :( As I saw on their webpage, this mailing list can be posted for this concern as well. I am using version 0.9.3 and acc. the installation file on http://www.openxcap.org all current dependencies from the Debian unstable distribution. When I start /usr/bin/openxcap, the following error occurs in the syslog: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema As I wrote, I followed all instructions from the openxcap webpage but didn't find any hints how to make it work. Any help on that would be appreciated. Best regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk <http://www.t-com.sk/>

4 3

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 10 Dec '07

10 Dec '07

Hello, can you please send unified format for the patch? diff -u oldfile newfile Thanks, Daniel On 12/10/07 14:29, Mihail Gluhowchenko wrote: > Hello; > Here it is sc.patch; > my version is 'openser-1.2.2-notls' > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX. > > 2007/12/10, Daniel-Constantin Mierla < daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>>: > > > > On 12/10/07 07:04, Mihail Gluhowchenko wrote: > > Hello all work; :) > was your version old or there is a bug in sc.dbtext? Can you post the > patch of changes? > > Best regards, > Daniel > > > > in script sc.dbtext use old format subscriber. > > Little changes and all work > > > > date_now=$(date +"%s") > > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | > sed -e > > 's/ //') > > > > add) > > if [ $# -ne 4 ] ; then > > usage > > exit 1 > > fi > > shift > > credentials $1 $2 > > is_user $1 > > if [ $? -eq 0 ] ; then > > echo -e "\nuser already exists\n" > > exit 1 > > fi > > #Add to SUBSCRIBER file > > #id(int,auto) username(str) domain(str) > password(str) > > first_name(str,null) last_name(str,null) email_address(str) > > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > > rpid(str,null) > > > > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > > > echo $line >> $SUB_FILE > > #Add to URI file > > #id(int,auto) domain(str) uri_user(str) > > last_modified(int) > > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > > echo $line >> $URI_FILE > > echo -e "\nNew user ($SERUSER@$SERDOMAIN) added to > > DBTEXT files\n" > > ;; > > -- > > Regards, > > Mihail Gluhowchenko. > > Enterprise ELTEX. > > > >

1 0

[Serusers] Problem for compiling with LCR
by Gerson A. Matiolli 10 Dec '07

10 Dec '07

Hi all I am compiling the SER-0.96 that downloaded the cvs with following command: make include_modules="mysql lcr" modules The following error occurred: make[1]: Entrando no diretÃ³rio `/home/ser/sip_router/modules/mysql' make[1]: `mysql.so' estÃ¡ atualizado. make[1]: Saindo do diretÃ³rio `/home/ser/sip_router/modules/mysql' make[1]: Entrando no diretÃ³rio `/home/ser/sip_router/modules/lcr' 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-dev13"' -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 -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 lcr_mod.c -o lcr_mod.o In file included from ../mysql/dbase.h:42, from lcr_mod.c:53: ../mysql/../../db/db_val.h:40: error: redeclaration of enumerator âDB_INTâ ../../db/db_fld.h:47: error: previous definition of âDB_INTâ was here ../mysql/../../db/db_val.h:41: error: redeclaration of enumerator âDB_FLOATâ ../../db/db_fld.h:48: error: previous definition of âDB_FLOATâ was here ../mysql/../../db/db_val.h:42: error: redeclaration of enumerator âDB_DOUBLEâ ../../db/db_fld.h:49: error: previous definition of âDB_DOUBLEâ was here ../mysql/../../db/db_val.h:44: error: redeclaration of enumerator âDB_STRâ ../../db/db_fld.h:51: error: previous definition of âDB_STRâ was here ../mysql/../../db/db_val.h:45: error: redeclaration of enumerator âDB_DATETIMEâ ../../db/db_fld.h:52: error: previous definition of âDB_DATETIMEâ was here ../mysql/../../db/db_val.h:46: error: redeclaration of enumerator âDB_BLOBâ ../../db/db_fld.h:53: error: previous definition of âDB_BLOBâ was here ../mysql/../../db/db_val.h:48: error: redeclaration of enumerator âDB_BITMAPâ ../../db/db_fld.h:55: error: previous definition of âDB_BITMAPâ was here lcr_mod.c:118: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âlcr_dbfâ lcr_mod.c: In function âlcr_db_initâ: lcr_mod.c:242: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:242: error: (Each undeclared identifier is reported only once lcr_mod.c:242: error: for each function it appears in.) lcr_mod.c: In function âlcr_db_bindâ: lcr_mod.c:260: warning: implicit declaration of function âbind_dbmodâ lcr_mod.c:260: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:266: warning: implicit declaration of function âDB_CAPABILITYâ lcr_mod.c:266: error: âDB_CAP_QUERYâ undeclared (first use in this function) lcr_mod.c:272: error: âDB_CAP_RAW_QUERYâ undeclared (first use in this function) lcr_mod.c: In function âlcr_db_closeâ: lcr_mod.c:284: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c: In function âlcr_db_verâ: lcr_mod.c:296: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:305: warning: implicit declaration of function âtable_versionâ lcr_mod.c: In function âreload_gwsâ: lcr_mod.c:503: error: âdb_row_tâ undeclared (first use in this function) lcr_mod.c:503: error: ârowâ undeclared (first use in this function) lcr_mod.c:518: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:534: warning: implicit declaration of function âRES_ROW_Nâ lcr_mod.c:542: warning: implicit declaration of function âRES_ROWSâ lcr_mod.c:543: warning: implicit declaration of function âROW_VALUESâ lcr_mod.c:543: error: invalid type argument of â->â lcr_mod.c:549: error: invalid type argument of â->â lcr_mod.c:550: error: invalid type argument of â->â lcr_mod.c:553: error: invalid type argument of â->â lcr_mod.c:561: error: invalid type argument of â->â lcr_mod.c:564: error: invalid type argument of â->â lcr_mod.c:572: error: invalid type argument of â->â lcr_mod.c:575: error: invalid type argument of â->â lcr_mod.c:584: error: invalid type argument of â->â lcr_mod.c:588: error: invalid type argument of â->â lcr_mod.c: In function âload_gwsâ: lcr_mod.c:636: error: âdb_row_tâ undeclared (first use in this function) lcr_mod.c:636: error: ârowâ undeclared (first use in this function) lcr_mod.c:636: error: ârâ undeclared (first use in this function) lcr_mod.c:636: warning: left-hand operand of comma expression has no effect lcr_mod.c:698: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:705: error: invalid type argument of â->â lcr_mod.c:709: error: invalid type argument of â->â lcr_mod.c:712: error: invalid type argument of â->â lcr_mod.c:714: error: invalid type argument of â->â lcr_mod.c:717: error: invalid type argument of â->â lcr_mod.c:719: error: invalid type argument of â->â lcr_mod.c:722: error: invalid type argument of â->â lcr_mod.c:724: error: invalid type argument of â->â lcr_mod.c:727: error: invalid type argument of â->â lcr_mod.c:729: error: invalid type argument of â->â lcr_mod.c:733: error: invalid type argument of â->â make[1]: ** [lcr_mod.o] Erro 1 make[1]: Saindo do diretÃ³rio `/home/ser/sip_router/modules/lcr' make: ** [modules] Erro 1 Somebody help me Tanks

1 0

[Serusers] Loading attributes question
by Steven C. Blair 10 Dec '07

10 Dec '07

I have a locally defined table with attributes which may contain several values. I define each attribute /value pair as a new entry in the table then use avp_load to retrieve these values from the database. The problem is that I don't know how to retrieve the values in a specific order. For example if I have a=1, a=2, a=3 when I load the values for "a" they may be 2,1,3 or 3,1,2, etc. Is there any way to force loading in a specific sequence. Thanks,Steve

1 0

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 10 Dec '07

10 Dec '07

On 12/10/07 07:04, Mihail Gluhowchenko wrote: > Hello all work; :) was your version old or there is a bug in sc.dbtext? Can you post the patch of changes? Best regards, Daniel > in script sc.dbtext use old format subscriber. > Little changes and all work > > date_now=$(date +"%s") > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | sed -e > 's/ //') > > add) > if [ $# -ne 4 ] ; then > usage > exit 1 > fi > shift > credentials $1 $2 > is_user $1 > if [ $? -eq 0 ] ; then > echo -e "\nuser already exists\n" > exit 1 > fi > #Add to SUBSCRIBER file > #id(int,auto) username(str) domain(str) password(str) > first_name(str,null) last_name(str,null) email_address(str) > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > rpid(str,null) > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > echo $line >> $SUB_FILE > #Add to URI file > #id(int,auto) domain(str) uri_user(str) > last_modified(int) > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > echo $line >> $URI_FILE > echo -e "\nNew user ($SERUSER@$SERDOMAIN) added to > DBTEXT files\n" > ;; > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX. >

1 0

[OpenSER-Users] Server Crash - Sending 500
by Kelvin Williams 10 Dec '07

10 Dec '07

While simulating a database failure (stopping the service), I noticed that OpenSER would also crash (see log below). While I don't mind the OpenSER crash since in our configuration its highly dependent upon the database, what I would like to do is send a 5xx series message to the client so that it would be forced to look at one of our other proxies to complete its call. As the client continued to send the INVITE to the original server. How can I send a 500 Server Internal Error message back to the client in the event of a database failure? Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: Probable invalid query Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: PGRES_FATAL_ERROR Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: server closed the connection unexpectedly This probably means the server terminated abnormally before or while processing the request. Dec 8 22:36:11 terra /usr/local/sbin/openser[8147]: ERROR: receive_fd: EOF on 23 Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: child process 8134 exited by a signal 11 Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: core was not generated Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: INFO: terminating due to SIGCHLD Dec 8 22:36:11 terra /usr/local/sbin/openser[8147]: INFO: signal 15 received Dec 8 22:36:11 terra /usr/local/sbin/openser[8140]: INFO: signal 15 received Dec 8 22:36:11 terra /usr/local/sbin/openser[8137]: INFO: signal 15 received

3 2

[OpenSER-Users] Timeout between 100 and 180/183 Messages
by Dioris Moreno 10 Dec '07

10 Dec '07

I know fr_timer controls the timeout for provisional responses, and fr_inv_timer controls the timeout for final responses. Also, the TM Module documentation explains: "fr_timer - this timer is used when no response was received yet. If there is no response after fr_timer seconds the timer triggers (and failure route will be executed if t_on_failure() was called). If a provisional response was received, the timer is set to fr_inv_timer for INVITE transactions, and RT_T2 for all other transactions. If a final response is received, the transaction has finished." I understand this "If a provisional response was received, the timer is set to fr_inv_timer for INVITE", but how can I set a different timeout for cases when I have received 100 Trying and I didn't get a 180/183 Ringing? This is the example mentioned in documentation: "For example: You want to have fail over if there is no provisional response after 3 seconds, but you want to ring for 60 seconds. Thus, set the fr_timer to 3 and fr_inv_timer to 60." How can I get a fail over if there is no provisional response after 3 seconds, a fail over if there is no 180/183 message after receiving 100 Trying of 10 seconds, and I want the SIP client rings for only 60 seconds? I'll appreciate any help Dioris Moreno

2 1

Re: [Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 10 Dec '07

10 Dec '07

To All, I offer comments / suggestion from perspective of enterprise user - more precisely from perspective of the IT-folks who will be installing and operating on behalf of their own enterprise. First, no matter what we select, we will not be able to satisfy those whose policy (or learning limitations) restrict them to a specific OS / DISTRO / VERSION / VENDOR. Just as there are myriads of enterprise with unique approaches to their particular industry, so also most have a very specific opinion about what works best for them. Therefore, I suggest we boldly assert, that SER is to be promoted as an appliance where the interests of the harried IT-folks can be best served. Let this bundle be delivered as a as a stand alone CD/DVD that, once it is deployed on a "buy it anywhere" machine, the IT-folks will never need to go beyond the external web / administrator interface into the inner realms of OS dependencies. <DREAM> This may be idealistic! However, broad scale enterprise acceptance can be achieved very quickly when we are able to achieve this goal. Oh, by-the-way, this broad user base who love the basic appliance for it's simplicity will also love those whose talents can deliver upscale versions of the same fundamental appliance. As one with enough grey hair and grandchildren to have loved my 195x vintage black and white television, I absolutely love the HD flat-screens of 2007! </DREAM> IMHO we should focus on an Enterprise level DISTRO. I agree with the statement "... I think the ultimate goal should be something like an ISO that's ready to go on a server..." Furthermore, I think virtual machine distributions are just another in a long line of "Silver Bullets" that may appeal to the developer heart in all of us . . . but which will scare the crap out of the policy focused Enterprise IT-folks. Therefore, I feel there is enough talent among us to create the first bundle in such a way to make it available to multiple Enterprise users so long as they understand it as an appliance! I have stated my preference for the CentOS-RHEL-FCx environment because it has the broadest base of enterprise-quality custom applications which equire mission-critical long-term stability. On the other hand, I will be supportive of the bundled appliance concept even if someone produces a VM - CentOS or another variation on the same theme. ..mike..

2 1

[Serusers] SER and Broadsoft
by Arun Kumar 10 Dec '07

10 Dec '07

Hi, I'm trying to configure my ser with broadsoft but to send calls to broadsoft it need to sent via a user no ip based auth. so, is there any why that I can send username/pass from ser to send calls to my broadsoft. thanks arun

2 2

[Serusers] Is SER for me ?
by Charalampos Alexopoulos 09 Dec '07

09 Dec '07

Hi Sorry if this is a stupid question, but before i dive to SIP protocol and SER configuration i would like to know if SER doing what i need. What i want to do is to make voip phone calls with my friends without to use some voip provider like skype. Is the SIP the correct think for this ? Is the SER implementation suitable ? Thank you in advance Charalampos Alexopoulos

3 2

[OpenSER-Users] configure openser with cpl module
by Martin Kardi 08 Dec '07

08 Dec '07

Hi, I installed openser with cpl module and other dependencies but I don't know how to configure it? Openser is functionally without cpl module. I am beginner in this. Could somebody help me how to? thanks Martin -- View this message in context: http://www.nabble.com/configure-openser-with-cpl-module-tf4942775.html#a141… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[OpenSER-Users] loose_route not setting $du ??
by Patrick Baker 07 Dec '07

07 Dec '07

I'm having an issue with messages flowing between asterisk and openser and I believe the issue is related to loose_route and having nulls for $du $dd $ds. Does anyone know how to resolve this with my config? I have also referenced my asterisk config along with some debug information. Thanks in advance for anyone that can help! Asterisk ends up dropping the call after 20 seconds as it appears that openSer isnt responding to asterisk OK message. Also BYE's and others arent working as well.. [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet. Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115' Method: INVITE ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Fri Dec 07 05:52:34 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Fri Dec 07 05:52:34 +0100 2007' ######################################################################## listen = udp:127.0.0.1:5060 listen = udp:10.3.1.31:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = no check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) { xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } t_on_failure("1"); route(3); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); } ##asterisk sip.conf## [general] matchexterniplocally=yes canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5061 bindaddr=0.0.0.0 sipdebug=yes nat=yes [openser] type=friend context=default insecure=very externalnotify=yes allow=all ##ser log## Dec 7 04:04:52 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.31:5061;transport=udp F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:52 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:53 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:54 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:56 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:00 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:04 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:08 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:12 phonesys-slave openser[24589]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:13 phonesys-slave openser[24599]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:16 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 ##asterisk sip debug## <--- SIP read from 10.3.1.31:5060 ---> INVITE sip:500@10.3.1.31:5061;transport=udp SIP/2.0 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31> Contact: <sip:pbaker2@10.3.1.115:5060> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE Max-Forwards: 69 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 252 v=0 o=pbaker2 3001829617 3001829777 IN IP4 10.3.1.115 s=X-Lite c=IN IP4 10.3.1.31 t=0 0 m=audio 35128 RTP/AVP 3 97 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=nortpproxy:yes <-------------> --- (13 headers 12 lines) --- Sending to 10.3.1.31 : 5060 (no NAT) Using INVITE request as basis request - 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 No user 'pbaker2' in SIP users list Found peer 'openser' for 'pbaker2' from 10.3.1.31:5060 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.1.31:35128 Found audio description format gsm for ID 3 Found audio description format speex for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x202 (gsm|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x202 (gsm|speex) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.3.1.31:35128 Looking for 500 in default (domain 10.3.1.31) list_route: hop: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> <--- Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Length: 0 <------------> Audio is at 167.206.216.136 port 37712 Adding codec 0x2 (gsm) to SDP Adding codec 0x200 (speex) to SDP Adding non-codec 0x1 (telephone-event) to SDP phonesys-slave*CLI> <--- Reliably Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 04:45:50] NOTICE[25642]: rtp.c:998 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.3.1.31 Retransmitting #6 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet. Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115' Method: INVITE

2 2

[OpenSER-Users] starting openser trunk version
by Daniel 07 Dec '07

07 Dec '07

Hi all, I'm using the latest trunk version of openser. When I start the proxy I see on debug this messages repeated forever... I'm using a config file that worked well wiht openser release 1.2.2. Dec 6 15:55:14 [28924] DBG:core:qm_free: params(0x817ada0, 0x81caa24), called from nathelper.c: nh_timer(2941) Dec 6 15:55:14 [28924] DBG:core:qm_free: freeing frag. 0x81caa0c alloc'ed from nathelper.c: nh_timer(2841) Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), called from nathelper.c: nh_timer(2841) Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), returns address 0x81caa24 frag. 0x81caa0c (size=8) on 1 -th hit Can you help me? what does it mean? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

4 3

[Serusers] SER From Header
by Arun Kumar 07 Dec '07

07 Dec '07

Hi, I've one exten on my ser box and one exten on Broadsoft switch. From SER I'm not able to dial my broadsoft exten, b'coz broadsoft expect known user in >From filed but I'm calling from SER so the From filed shows my SER user not my broadsoft. So how do I change my From header when I call my broadsoft. Im using ser-0.9.6 version. Please help. Thanks, Arun

1 0

[OpenSER-Users] Accounting issues
by David Villasmil Govea 07 Dec '07

07 Dec '07

Hello all, I finally got accounting on db to work. I added a few extra fields and its all fine. But i have some questions: I added: modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") to me config, and all the info gets loaded. BUT, source_ip is not always the ip from which the request was received, either that, or INVITES are not getting logged! well, that's really 2 questions in one... ;) here's my config: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=20 disable_dns_blacklist=true # # --- module loading mpath="/usr/lib/openser/modules/" 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 "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" #loadmodule "domain.so" loadmodule "xlog.so" loadmodule "/usr/lib/openser/modules/acc.so" modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_url", "mysql://openser:password@localhost/openser") modparam("acc", "report_ack", 0) modparam("acc", "db_missed_flag", 3) modparam("acc", "failed_transaction_flag", 4) modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # # CONSTANTS # # --- setting module parameters #modparam("domain","db_url","mysql://openser:password@localhost/openser") #modparam("domain", "db_mode", 1) #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") modparam("xlog", "buf_size", 8192) # --- setting module parameters modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc|auth_db","db_url","mysql://openser:password@localhost /openser") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "nat_bflag", 6) #modparam("usrloc", "use_domain", 0) # -- registrar params -- modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("auth_db", "use_domain", 0) # -- rr params -- #modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:1.2.3.4:8899") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@sip.whatever.com") #modparam("registrar", "max_contacts", 1) # --- main routing logic route{ if (method=="INVITE") { xlog("L_NOTICE", "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(2); setflag(3); setflag(4); }; if (method=="BYE") { xlog ("L_NOTICE", "BYE - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="CANCEL") { xlog ("L_NOTICE", "CANCEL - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="REGISTER") { setflag(2); setflag(3); setflag(4); }; if (!mf_process_maxfwd_header("100")) { sl_send_reply("483","Too Many Hops"); exit; }; #if (msg:len >= 2048 ) { # sl_send_reply("513", "Message too big"); # exit; #}; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # if (!is_uri_host_local() ) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_NOTICE", "\nNew Registration\n"); if (!www_authorize("sip.whatever.com", "subscriber")) { xlog("L_NOTICE", "\nRegistration FAILED! CLIENT: $au\n"); www_challenge("sip.whatever.com", "0"); exit; }; if (isflagset(5)) { xlog("L_NOTICE", "\nRegistration OK\n"); setbflag(6); # if you want OPTIONS natpings uncomment next # setbflag(7); }; save("location"); exit; }; }; route(1); } route[1] { xlog("L_NOTICE", "\n**************IN ROUTE 1************\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } #======================================= Thanks in advance! Any other comment to make this cfg better, i'm all ears!! David

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[OpenSER-Users] Make uninstall
by Daniel 07 Dec '07

07 Dec '07

Hi all, I have to uninstall openser from my pc. Is there a "make uninstall" ? Regerds, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Problem with Mediaproxy
by Daniel 07 Dec '07

07 Dec '07

Dear all, I'm testing my proxy and I'm using mediaproxy for natted client. Everything works fine, if there is a client behind nat calling a public user, mediaproxy works fine. This is true also if there is a public client calling a natted user. I have some problem when there are 2 client behind NAT. 1. I tryed to test the system with 1 client behind NAT and 1 client behind the SAME NAT (both of client under the same subnet:192.168.10.x). In this case Mediaproxy doesn't work. RTP in direct between clients. 2. I tryed to test the system with 1 client behind NAT and 1 client behind the SAME NAT (with clients under different subnet:192.168.10.x and 192.168.0.x). In this case Mediaproxy doesn't work. RTP in direct between clients. This is Mediaproxy Debug in the last case: ********************************************** Dec 7 10:37:36 SERVER mediaproxy[2116]: request ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI. 192.168.0.167:58786:audio NAT_IP PROXY_IP remote NAT_IP remote X-Lite=20release=201011s=20stamp=2041150 info=from:userA@PROXY_IP, to:UserB@PROXY_IP,fromtag:c1310701,totag: Dec 7 10:37:36 SERVER mediaproxy[2116]: session ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI.: started. listening on MEDIAPROXY_IP:60014 Dec 7 10:37:36 SERVER mediaproxy[2116]: execution time: 1.56 ms Dec 7 10:37:43 SERVER mediaproxy[2116]: lookup ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI. 192.168.10.224:25732:audio NAT_IP PROXY_IP remote unknown unknown X-Lite=20release=201011s=20stamp=2041150 info=from:UserA@PROXY_IP, to:UserB@PROXYIP,fromtag:c1310701,totag:67246350 Dec 7 10:37:43 SERVER mediaproxy[2116]: execution time: 0.51 ms Dec 7 10:38:00 SERVER mediaproxy[2116]: delete ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI. info= Dec 7 10:38:00 SERVER mediaproxy[2116]: session ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI.: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Dec 7 10:38:00 SERVER mediaproxy[2116]: session ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI.: ended. Dec 7 10:38:00 SERVER mediaproxy[2116]: execution time: 0.61 ms ******************************* I am sure of mediaproxy change SDP parameter even in these last cases (I can see on openser debug that proxy send out, for example, a 200 OK with SDP part changed. And xlogs DEBUG show me that openser enters in "use_media_proxy" routine) . But when I sniff the packet on client side, I see SDP part with private IP ! Any suggestion? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] T.38 Fax Problem with RTPProxy
by Raymond Chandler 07 Dec '07

07 Dec '07

In the following, I have a T.38 Fax call in which I'm forcing RTPProxy. The signalling follows this draft (http://tools.ietf.org/html/draft-mule-sip-t38callflows-02#section-5.2), but the UDPTL media seems to have problems. The problem is that OpenSER/RTPProxy seems to ignore the address in the SDP of the "200 OK". As I understand it, RTPProxy should, adequately enough, proxy the UDPTL. (not just rtp) The call flow is as follows: UC -> EDGE EDGE -> OpenSER OpenSER -> EU (not shown in this trace) NOTE: IPs have been changed to protect the innocent ;-) 192.168.174.156 represents our UC's (Underlying Carrier's) IP 192.168.48.88 represents our edge proxy 192.168.202.10 represents our OpenSER box Below, I have what I consider to be the "relevent" packets with large portions of repetition SNIP'd.... Below that, I have the two (again, what I consider to be) relevent packets with details of the SDP In both of the SDP details, our edge uses the UC's IP (192.168.174.156) in the c= line of the SDP. I don't see that being overwritten in any of the "media-specific" sections following. OpenSER/RTPProxy handle the RTP appropriately (to the best of my knowledge) but send the t.38 UDPTL media back to our edge which is, of course, rejecting it. "No.", "Time", "Source", "Destination", "Protocol", "Info" "1", "0.000000", "192.168.48.88", "192.168.202.10", "SIP/SDP", "Request: INVITE sip:1NPANXX0119@192.168.202.10:6080, with session description" "2", "0.005027", "192.168.202.10", "192.168.48.88", "SIP", "Status: 100 Trying" "3", "0.286018", "192.168.202.10", "192.168.48.88", "SIP", "Status: 180 Ringing" "4", "9.397591", "192.168.202.10", "192.168.48.88", "SIP/SDP", "Status: 200 OK, with session description" "5", "9.410633", "192.168.202.10", "192.168.48.88", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3122798450, Seq=5169, Time=374804094" "6", "9.439195", "192.168.48.88", "192.168.202.10", "SIP", "Request: ACK sip:1NPANXX0119@192.168.202.10:6080" "7", "9.440987", "192.168.202.10", "192.168.48.88", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3122798450, Seq=5170, Time=374804334" "8", "9.448560", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "9", "9.470710", "192.168.202.10", "192.168.48.88", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3122798450, Seq=5171, Time=374804574" "10","9.479789","69.25.48.88","67.15.202.10","ICMP","Destination unreachable (Port unreachable)" "11","9.490158","207.218.174.156","67.15.202.10","RTP","PT=ITU-T G.711 PCMU, SSRC=0xCF806328, Seq=0, Time=0, Mark " "12","9.500186","67.15.202.10","207.218.174.156","RTP","PT=ITU-T G.711 PCMU, SSRC=0xBA221F72, Seq=5172, Time=374804814 " "13","9.509146","69.25.48.88","67.15.202.10","ICMP","Destination unreachable (Port unreachable)" "14","9.510270","207.218.174.156","67.15.202.10","RTP","PT=ITU-T G.711 PCMU, SSRC=0xCF806328, Seq=1, Time=160 " "15","9.530265","207.218.174.156","67.15.202.10","RTP","PT=ITU-T G.711 PCMU, SSRC=0xCF806328, Seq=2, Time=320 " "16","9.530283","67.15.202.10","207.218.174.156","RTP","PT=ITU-T G.711 PCMU, SSRC=0xBA221F72, Seq=5173, Time=374805054 " ~~~~ SNIP ~~~~ "174", "11.430039", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=97, Time=15520" "175", "11.450027", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=98, Time=15680" "176", "11.457118", "192.168.202.10", "192.168.48.88", "SIP/SDP", "Request: INVITE sip:Restricted@192.168.48.88, with session description" "177", "11.468025", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "178", "11.470013", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=99, Time=15840" "179", "11.490125", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=100, Time=16000" "180", "11.492010", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "181", "11.499118", "192.168.48.88", "192.168.202.10", "SIP", "Status: 100 Trying" "182", "11.510111", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=101, Time=16160" "183", "11.515999", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "184", "11.530100", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=102, Time=16320" "185", "11.540998", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "186", "11.550087", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=103, Time=16480" "187", "11.570073", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=104, Time=16640" "188", "11.580068", "192.168.174.156", "192.168.202.10", "RTP", "Unknown RTP version 0" "189", "11.581986", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "190", "11.590061", "192.168.174.156", "192.168.202.10", "RTP", "Unknown RTP version 0" "191", "11.599929", "192.168.174.156", "192.168.202.10", "RTP", "Unknown RTP version 0" "192", "11.605977", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "193", "11.608549", "192.168.48.88", "192.168.202.10", "SIP/SDP", "Status: 200 OK, with session description" "194", "11.630936", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "195", "11.660168", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "196", "11.668760", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "197", "11.690405", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "198", "11.697992", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "199", "11.720762", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "200", "11.728847", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "201", "11.750120", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "202", "11.758827", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "203", "11.780601", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 t30ind: no-signal hdlc-fcs-BAD-sig-end t4-non-ecm-sig-end [Malformed?] hdlc-fcs-BAD-sig-end hdlc-data[Malformed Packet]" "204", "11.788059", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "205", "11.811073", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 t30ind: no-signal hdlc-fcs-BAD-sig-end t4-non-ecm-sig-end [Malformed?] hdlc-fcs-BAD-sig-end hdlc-sig-end[Malformed Packet]" "206", "11.818665", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "207", "11.833039", "192.168.202.10", "192.168.48.88", "SIP", "Request: ACK sip:Restricted@192.168.48.88:5060" "208", "11.839935", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 t30ind: cng hdlc-fcs-BAD-sig-end t4-non-ecm-sig-end [Malformed?] hdlc-fcs-BAD-sig-end hdlc-fcs-OK[Malformed Packet]" "209", "11.849145", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "210", "11.868409", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=00000 t30ind: no-signal" "211", "11.878000", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "212", "11.905983", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "213", "14.428661", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=00001 t30ind: v21-preamble" ~~~~ SNIP ~~~~ "389", "32.869453", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=00114 t30ind: no-signal" "390", "41.696978", "192.168.202.10", "192.168.48.88", "SIP", "Request: BYE sip:Restricted@192.168.48.88:5060" "391", "41.738094", "192.168.48.88", "192.168.202.10", "SIP", "Status: 200 OK" No. Time Source Destination Protocol Info 1 0.000000 192.168.48.88 192.168.202.10 SIP/SDP Request: INVITE sip:1NPANXX0119@192.168.202.10:6080, with session description Frame 1 (966 bytes on wire, 966 bytes captured) Ethernet II, Src: 00:1e:13:ca:4a:3f (00:1e:13:ca:4a:3f), Dst: Dell_ed:4a:1c (00:12:3f:ed:4a:1c) Internet Protocol, Src: 192.168.48.88 (192.168.48.88), Dst: 192.168.202.10 (192.168.202.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 6080 (6080) Session Initiation Protocol Request-Line: INVITE sip:1NPANXX0119@192.168.202.10:6080 SIP/2.0 Message Header Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): BRSDP 187392 187392 IN IP4 192.168.174.156 Session Name (s): BRSDP Session Connection Information (c): IN IP4 192.168.174.156 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 53278 RTP/AVP 0 8 18 100 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:100 telephone-event/8000 Media Attribute (a): fmtp:100 0-15 No. Time Source Destination Protocol Info 193 11.608549 192.168.48.88 192.168.202.10 SIP/SDP Status: 200 OK, with session description Frame 193 (1013 bytes on wire, 1013 bytes captured) Ethernet II, Src: 00:1e:13:ca:4a:3f (00:1e:13:ca:4a:3f), Dst: Dell_ed:4a:1c (00:12:3f:ed:4a:1c) Internet Protocol, Src: 192.168.48.88 (192.168.48.88), Dst: 192.168.202.10 (192.168.202.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 6080 (6080) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): BRSDP 187392 187393 IN IP4 192.168.174.156 Session Name (s): BRSDP Session Connection Information (c): IN IP4 192.168.174.156 Time Description, active time (t): 0 0 Media Description, name and address (m): image 53278 udptl t38 Media Attribute (a): T38FaxVersion:0 Media Attribute (a): T38MaxBitRate:14400 Media Attribute (a): T38FaxRateManagement:transferredTCF Media Attribute (a): T38FaxMaxBuffer:262 Media Attribute (a): T38FaxMaxDatagram:176 Media Attribute (a): T38FaxUdpEC:t38UDPRedundancy

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[OpenSER-Users] Does MediaProxy have to be loaded if there is no NAT?
by Brian Heath 07 Dec '07

07 Dec '07

So I'm trying to diagnose an intermittent bug we seem to be having, and I'm not sure if Mediaproxy's the cause or not... When a call is made through our system, and then the call is ended normally, the logs show the following mediaproxy entries: Dec 6 12:14:02 server01 mediaproxy[14777]: delete 7dfafbf931195f8b61d7bbea68f9c0f8(a)xxx.xxx.xxx.xx1 info= Dec 6 12:14:02 server01 mediaproxy[14777]: execution time: 0.20 ms It seems to be pretty standard stuff, really. However, if a call is not made from behind a firewall, why is mediaproxy even running? Alas, this is not my main problem... Every once in a while, calls get dropped. It happens more frequently than is acceptable. One common thing I've noticed when this happens, is mediaproxy will log a 'request' and start a session, and then approximately 30 seconds later, the call will be terminated. Take this one for example. A customer originated a call this morning at Dec 6 11:25:18. Half an hour later, mediaproxy terminated the call. This was the log entry: Dec 6 11:55:37 server01 mediaproxy[14777]: request 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1 xxx.xxx.xxx.xx2:10508:audio xxx.xxx.xxx.xx2 sip.mydomain.com remote xxx.xxx.xxx.xx1 remote Quintum/1.0.0 info=from:1925XXXXXXX@sip.mydomain.com,to:1925XXXXXXX@xxx.xxx.xxx.xx1,fromtag:45364b32-2439a9,totag:as694a235d Dec 6 11:55:37 server01 mediaproxy[14777]: session 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1: started. listening on xxx.xxx.xxx.xx3:60490 Dec 6 11:55:37 server01 mediaproxy[14777]: execution time: 1.12 ms Dec 6 11:56:08 server01 mediaproxy[14777]: delete 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1 info= Dec 6 11:56:08 server01 mediaproxy[14777]: session 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Dec 6 11:56:08 server01 mediaproxy[14777]: session 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1: ended. Dec 6 11:56:08 server01 mediaproxy[14777]: execution time: 0.68 ms Is there any reason why mediaproxy might be arbitrarily loading and starting a new session, when the call has already long been established? I can't seem to make heads or tails of this. I'd appreciate any comments or suggestions. Thanks, Brian This question is also posted at: http://www.voipuser.org/forum_topic_11777.html

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Re: [OpenSER-Users] starting openser trunk version
by Daniel 06 Dec '07

06 Dec '07

On Thursday 06 December 2007, Daniel wrote: > I'm using the latest trunk version of openser. > When I start the proxy I see on debug this messages repeated forever... > I'm using a config file that worked well wiht openser release 1.2.2. > > Dec 6 15:55:14 [28924] DBG:core:qm_free: params(0x817ada0, 0x81caa24), > called from nathelper.c: nh_timer(2941) > Dec 6 15:55:14 [28924] DBG:core:qm_free: freeing frag. 0x81caa0c alloc'ed > from nathelper.c: nh_timer(2841) > Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), called > from nathelper.c: nh_timer(2841) > Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), returns Hi Daniel, this is the (harmless) memory debugging that was activated to catch some errors in memory management. This will be decativated next week, when 1.3 is released. So you can either deactivate it by yourself (look in Makefile.defs), ignore the messages or wait one week. ;-) Cheers, Henning Hi Henning, but now how can I see the classic debug, without these messages scrolling on the screen? Thanks again, Regards Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] OpeSer SVN presence error
by TCB 06 Dec '07

06 Dec '07

I'm trying to setup openser-svn-3261 with presence support and gettign the following error. Dec 5 18:04:27 ss-za /usr/local/sbin/openser[7403]: Handle PUBLISH Dec 5 18:04:27 ss-za /usr/local/sbin/openser[7403]: ERROR:presence:handle_publish: Missing or unsupported event header field value Dec 5 18:04:27 ss-za /usr/local/sbin/openser[7403]: ERROR:presence:handle_publish: event=[presence I'm also having issues starting Oser with presence_xml module, its crashes with no error in my log. Disable it and oser start fine. -- TC

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[OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1 Released
by Dan-Cristian Bogos 06 Dec '07

06 Dec '07

Folks, for any of you using (or having interest in using) prepaid authentication based on CDRTool through FreeRADIUS Authorization methods, I announce the release 1.1 available to be downloaded from SourceForge.net. Changelog from version 1.0 to present: !Version 1.1 - 05.12.2007 by Dan-Cristian Bogos [dan(a)itsyscom.com] * Replaced MAXDUR setting with SESSION_MAXDUR to be more descriptive * Introducing experimental support for multiple cdrtool engines by setting MULTIPLE_PREPAID_ENGINES=True * Simplified way of testing memcached connection (simple test against response list). * Fixed BUG with sending non supported methods to CDRTool (eg: REGISTER and MESSAGE). * Automatic mark of authenticated packets by returning third sip-avp "authentic#1". * Return USERLOCK instead of REJECT when the account is locked. * Added authenticate() function to comply with freeradius internals - not really necessary to be used. * Fixed BUG which was failing accounting if Acct-Type was not fix positioned inside parameter tuple coming from FreeRADIUS. * Introduced Acct-Authentic attribute dependency in order to fix double accounting in case of Yate(Cisco CDR with two leg accounting type). OpenSER should send Acct-Authentic attribute as radius_extra param with the value 1 for RADIUS or 3 for Remote. * Fixed BUG which was emptying destination-uri in case of empty Canonical-URI. * Forcing timeout for the keys in memcache based on SESSION_MAXDUR, for the case when no more accounting packets will arive after authorizaiton ones. * Introduced support for Acct-Session-Time attribute which will produce more aqurate session time then automatic generated based on timestamps ones. * Introduced support for combined accounting to fix the problems OpenSER has when no BYE packet arrives (eg: authorization generated by openser and accounting by Yate). * Introduced support for newer cdrtool response 'Not Prepaid' derived from old 'NotPrepaid' * Proper detection in case of Stop packets without Start being recorded inside memcached. * Updated README-txt file to reflect the changes necessary in the configuration of OpenSER plus simplified way of installing freeradius with "./configure --with-experimental-modules" I am at your disposal if any question might pop-up. Project link: https://sourceforge.net/projects/frad-cdrtool DanB

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[Serusers] mailbox
by Szasz Szabolcs 06 Dec '07

06 Dec '07

Hello! I'm new in SEMS, I have a problem using mailbox application. I have set up mailbox to use the same imap mailbox to store the messages and mailbbox_query to get from there the messages, it works in this way. 1. We have two numbers: 200 and 201, respectivelly we have imap subfolders inbox/200/ and inbox/201/. 200 calls 201, 201 isn't online, sems records a message and I want to put in 201's imap subfulder this message. 2. when 201 is calling the mailbox_query I would like to be able to access direct the inbox/201 folder and get from there the messages. How can I configure to store in different subfolders for different destination numbers? How to get a number the mailbox messages from his subfolder when he calls in the mailbox_query? Thank you Szasz Szabolcs

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Re: [Serusers] myself value
by franz.edler＠inode.at 06 Dec '07

06 Dec '07

> So why do I receive icmp response on sip register request. If ser built > properly the myself value it should know that "new" > myselfxxx.xxx.xxx.xxxx:5066 and to take care or requests he receives on > port 5066 on sip level Two questions: 1. Are the ICMP responses generated by requests destined to port 5066? 2. Did you set the port directive accordingly in ser.cfg? Franz

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[Serusers] SIP Express Bundle ready-to-go idea, request for input
by Greger V. Teigre 06 Dec '07

06 Dec '07

Hi guys, I have been playing with the following idea: Create a ready-to-run OS image with everything that is needed for iptel.org apps pre-installed + a complete installation of: * SER 2.0 (release) * rtpproxy * SEMS * SERweb * maybe sipsak, some monitoring tools, etc The idea is to create a small script 'config_iptelorg' for configuring the installation to your needs. You should then be able to download the ready image, boot it, go through the script and have an up and running iptel.org proxy and app server just like the iptel.org free SIP service in maybe 10-15 minutes. This way you could host a SIP service for your own domain with close to no setup at all. Some questions to you: * Is there any interest for this at all? * I was thinking about using Ubuntu 7.10 server as the OS. Any thoughts/preferences? * Should the image be an Amazon EC3 image (you could use http://www.rightscale.com and get it running in no time with 10 run-hours free) or should it be a VMware appliance to be run with free VMware Player? * Other suggestions/comments? I could need some help with this, anyone interested in lending me a hand? (could be anything, documentation, testing, installation, etc) g-)

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[OpenSER-Users] force_tcp_alias
by Johnny Karlsen 06 Dec '07

06 Dec '07

There are numerous discussion on using force_tcp_alias() in order to route all SIP requests over an already established TCP between the client and OpenSER, however, I cannot make it work. What I would like to accomplish, any client registering using TCP should have all SIP requests sent towards the client using the already established TCP and not routed based on the Contact header used during REGISTER. Reading posts it would seem that putting force_tcp_alias(); within the route{ } should do the trick. However, this doesn't seem to do the trick for me, SIP requests towards the client are still routed using the Contact information and for example a SIP MESSAGE is sent over UDP from OpenSER towards the client using the information in the Conatct header. Any hints what to look for? Many thanks

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[Serusers] myself value
by Evgeny Miloslavsky 06 Dec '07

06 Dec '07

Hi I'm kind of newbie at all the SER issue so sorry for bothering you with my questions. I defined at ser.cfg to listen on port 5066 instead of default 5060 port, saved changes and restarted the SER. At Wireshark sniffer I see that there is an ICMP--> Destination Port Unreachable response on all sip REGISTER requests arrived from various endpoints at my setup. As far as I understand this means that configured 5066 port is not bound with SER application. So my question is: where can I see, how is "myself" built or concatenation of which parameters creates "myself" value? Thanks

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[OpenSER-Users] update trunk version
by Daniel 06 Dec '07

06 Dec '07

Hi, I'm trying to compile the latest trunk version of cpl-c module. I used the following commands: make make modules=modules/cpl-c modules but I had these errors: Compiling cpl.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"openser"' -DVERSION='"1.2.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -c cpl.c -o cpl.o cpl.c:52:24: error: ../../pvar.h: No such file or directory cpl.c:118: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cpl_procs' cpl.c:129: warning: excess elements in struct initializer cpl.c:129: warning: (near initialization for 'cmds[0]') cpl.c:131: warning: excess elements in struct initializer cpl.c:131: warning: (near initialization for 'cmds[1]') cpl.c:133: warning: excess elements in struct initializer cpl.c:133: warning: (near initialization for 'cmds[2]') cpl.c:134: warning: excess elements in struct initializer cpl.c:134: warning: (near initialization for 'cmds[3]') cpl.c:183: error: 'cpl_procs' undeclared here (not in a function) cpl.c:184: warning: initialization from incompatible pointer type cpl.c:185: warning: initialization from incompatible pointer type cpl.c:186: warning: initialization from incompatible pointer type cpl.c:188: warning: excess elements in struct initializer cpl.c:188: warning: (near initialization for 'exports') cpl.c: In function 'fixup_cpl_run_script': cpl.c:202: warning: implicit declaration of function 'LM_ERR' cpl.c: In function 'cpl_init': cpl.c:233: error: 'pv_spec_t' undeclared (first use in this function) cpl.c:233: error: (Each undeclared identifier is reported only once cpl.c:233: error: for each function it appears in.) cpl.c:233: error: expected ';' before 'avp_spec' cpl.c:237: warning: implicit declaration of function 'LM_INFO' cpl.c:241: warning: implicit declaration of function 'LM_CRIT' cpl.c:260: warning: implicit declaration of function 'pv_parse_spec' cpl.c:260: error: 'avp_spec' undeclared (first use in this function) cpl.c:261: error: 'PVT_AVP' undeclared (first use in this function) cpl.c:266: warning: implicit declaration of function 'pv_get_avp_name' cpl.c:353: warning: implicit declaration of function 'LM_NOTICE' make[1]: *** [cpl.o] Error 1 make[1]: Leaving directory `/usr/src/openser-1.2.2-notls/modules/cpl-c' make: *** [modules] Error 2 what's the problem? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] CPL Errors
by Daniel 05 Dec '07

05 Dec '07

Hi all, What does this error means ? It occurs only when I call a CPL script af a natted User. Refer to OUTGOING CPL Script: 4(9914) ERROR:cpl_c: overflow detected ip=0xb608fa30 offset=506 in func. cpl_run.c, line 1000 4(9914) ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established Refer to INCOMING CPL Script: 4(9914) ERROR:cpl_c:cpl_proxy_to_loc_set: do_action failed 4(9914) ERROR:cpl_c:cpl_run_script: runtime error Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Openser is giving compile errors for avp_write and avp_pushto
by Sandeep Ammaladinna 05 Dec '07

05 Dec '07

Hi Henning, I commented out the "fifo" part and still the config file errs. When i tried with openser -c option it says, config file OK. But when I actually run the openser it cribs. Attached is the log file. Cheers, -Sandeep -----Original Message----- From: users-bounces(a)lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf Of users-request(a)lists.openser.org Sent: Wednesday, December 05, 2007 5:30 PM To: users(a)lists.openser.org Subject: Users Digest, Vol 31, Issue 11 Send Users mailing list submissions to users(a)lists.openser.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.openser.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.openser.org You can reach the person managing the list at users-owner(a)lists.openser.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. configure openser with cpl module (Martin Kardi) 2. Re: Proxy with OpenSER (Andreas M.) 3. Openser is giving compile errors for avp_write and avp_pushto (Sandeep Ammaladinna) 4. Re: Openser is giving compile errors for avp_write and avp_pushto (Henning Westerholt) 5. Re: uac_replace_from issues (Henning Westerholt) 6. Could someone help me with WeSIP ? (Daniel Grotti) ---------------------------------------------------------------------- Message: 1 Date: Tue, 4 Dec 2007 04:15:18 -0800 (PST) From: Martin Kardi <martin.kardos(a)pego.sk> Subject: [OpenSER-Users] configure openser with cpl module To: users(a)lists.openser.org Message-ID: <14149601.post(a)talk.nabble.com> Content-Type: text/plain; charset=us-ascii Hi, I installed openser with cpl module and other dependencies but I don't know how to configure it? Openser is functionally without cpl module. I am beginner in this. Could somebody help me how to? thanks Martin -- View this message in context: http://www.nabble.com/configure-openser-with-cpl-module-tf4942775.html#a 14149601 Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. ------------------------------ Message: 2 Date: Wed, 05 Dec 2007 08:57:33 +0100 From: "Andreas M." <openser(a)breitenlee.net> Subject: Re: [OpenSER-Users] Proxy with OpenSER To: openser list user <users(a)lists.openser.org> Message-ID: <475659ED.9060809(a)breitenlee.net> Content-Type: text/plain; charset=ISO-8859-1 Hello, an IMS P-CSCF is not just a Proxy, there are a lot of routines and functions inside, that could not be done with an IETF SIP Proxy. Take a look at www.openimscore.org, there are also config examples, based on SER. r, Andreas M. hammou moujane schrieb: > Hello everybody, > > I am new in this mailing list and new user of OpenSER. I would like to > know where can I find the doc which can help me to make OpenSER as a > proxy server like P-CSCF of IMS core. what to be modified in > openser.conf configuration file. > > > > Bests Regards. > -- > Hammou MOUJANE > Tel: +212 62 46 33 89 > INPT > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users -- g, Andreas M. ------------------------------ Message: 3 Date: Wed, 5 Dec 2007 14:13:11 +0530 From: "Sandeep Ammaladinna" <sammaladinna(a)velankani.com> Subject: [OpenSER-Users] Openser is giving compile errors for avp_write and avp_pushto To: <users(a)lists.openser.org>, <devel(a)lists.openser.org> Message-ID: <001b01c8371a$e14dd170$1010000a(a)blr.velankani.com> Content-Type: text/plain; charset="us-ascii" Hi, When I am compiling my openser.cfg, Openser is throwing errors for avp_write and avp_pushto. The syntax I am using is as follows: avp_write("$ruri", "i:10"); avp_pushto("$ruri", "i:10"); What could be the issue; I am following the syntax from the cook book. Attached is my .cfg Thanks in advance. Cheers -Sandeep

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[OpenSER-Users] Openser is giving compile errors for avp_write and avp_pushto
by Sandeep Ammaladinna 05 Dec '07

05 Dec '07

Hi, When I am compiling my openser.cfg, Openser is throwing errors for avp_write and avp_pushto. The syntax I am using is as follows: avp_write("$ruri", "i:10"); avp_pushto("$ruri", "i:10"); What could be the issue; I am following the syntax from the cook book. Attached is my .cfg Thanks in advance. Cheers -Sandeep

3 2

[OpenSER-Users] uac_replace_from issues
by Alex Hermann 05 Dec '07

05 Dec '07

Hello, I have 2 issues with uac_replace_from. 1) Can someone add a big fat warning to the documentation that "append_fromtag" from the rr module MUST be enabled for automatic restore to function? I got random memory locations copied into the To: or From: header when it was disabled. 2) I get the message "ERROR:uac:restore_from: new URI shorter than old URI". Is there a limitation to what I can replace or what it can be replaced with? Can I safely remove this limitation in modules/uac/from.c ? diff -ru openser-1.2-orig/modules/uac/from.c openser-1.2/modules/uac/from.c --- openser-1.2-orig/modules/uac/from.c 2007-10-22 11:48:48.000000000 +0200 +++ openser-1.2/modules/uac/from.c 2007-12-04 17:20:44.000000000 +0100 @@ -472,10 +472,6 @@ } /* get new uri */ - if ( new_uri.len<old_uri.len ) { - LOG(L_ERR,"ERROR:uac:restore_from: new URI shorter than old URI\n"); - goto failed; - } for( i=0 ; i<old_uri.len ; i++ ) new_uri.s[i] ^= old_uri.s[i]; if (new_uri.len==old_uri.len) { if ( new_uri.len<old_uri.len ) { LOG(L_ERR,"ERROR:uac:restore_from: new URI shorter than old URI\n"); goto failed; } -- Greetings, Alex Hermann

2 1

[OpenSER-Users] Proxy with OpenSER
by hammou moujane 05 Dec '07

05 Dec '07

Hello everybody, I am new in this mailing list and new user of OpenSER. I would like to know where can I find the doc which can help me to make OpenSER as a proxy server like P-CSCF of IMS core. what to be modified in openser.conf configuration file. Bests Regards. -- Hammou MOUJANE Tel: +212 62 46 33 89 INPT

2 1

[OpenSER-Users] Parallel forking to loopback problem - for ringing multiple numbers at same time, simulring
by Dave Singer 05 Dec '07

05 Dec '07

by the way, I'm trying this using openser 1.3.0-pre1-notls from latest svn of trunk as of 12-4-07 I'm trying to do findme logic.. Call comes in, lookup settings for where it is supposed to go. >From the settings I find that the user wants it to ring a friends home phone, a cell phone, and the home ata associated with account. To keep voicemail handled by the home account and not by one of the places that they are forwarding to they want voicemail to be triggered after 15 seconds. The friends home phone is actually one of our accounts and has two ata's associated with it that register as the same sip user and they have it also ring a cell phone and the account sends the call to voicemail after 18 seconds. Psudo Call flow/Logic of what I have so far. Call comes in to openser for 555/123-4567 pull from usr_preferences settings for the call destination number. so I get a stack of avps $avp(s:brahch_uri) from usr_preferences like this: sip:5551239876@127.0.0.1 sip:localuser3@127.0.0.1 sip:5553216548@127.0.0.1 I do diversion header then branch for them with: avp_pushto("$br", "$avp(s:branch_uri)" I drop the local branch. The branched calls come in as expected and verify that it is a valid intentional loopback by testing: if ( ruri =~ "127.0.0.1" && dst_ip == 127.0.0.1 ) { < mark as valid loopback > } Like the beginning of the original call I lookup where they should go and if the user in the diversion header is allowed to send them there (PSTN) In this case one call would be sent out through the PSTN gateways with full failover logic, another would call the device associated with the account by doing a lookup("location") and the third, (the one to the friend that our customer) would add another diversion header and start another parallel fork with loopback for the cell phone and two ata's When I test with a simple forward to a single PSTN number, in the loopbacked session, I rewrite the ruri accordingly with real dest user and real ip address. the ruri shows what I want in the logs just before t_relay and exit and in branch_route[1] after last log entry from branch_route, I have in the log: ERROR:core:udp_send: sendto(sock,0xb619d954,1376,0,0xb619ba6c,16): Invalid argument(22) CRITICAL:core:udp_send: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net ERROR:tm:msg_send: udp_send failed ERROR:tm:t_forward_nonack: sending request failed Then I have log entrys from onreply_route[1]: Reply - S=100 M=INVITE RURI=<null> src_ip = 127.0.0.1 Reply - S=477 M=INVITE RURI=<null> src_ip = 127.0.0.1 not the real world IP address of the other server I set in the ruri and there is nothing in the logs on the intended server or the local server indicating routing logic for an invite failed I'm woundering if the TM module, while processing the invite in the loopback, is matching up the transaction with the session that originated the loopback and using the settings and status from it. Any Ideas, Thoughts, HELP!!? Or is OpenSER just not ready for this yet? Thanks Dave

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[OpenSER-Users] uac_restore_from problem
by Zoltán Kovács 05 Dec '07

05 Dec '07

Dear Colleagues, first of all, thank you for the great software which is a very nice and convenient way to build up a stable VoIP telephone system! We already have a working system based on OpenSER and Asterisk. After many experiments I finally raise this question to you because I cannot figure out the solution. The problem described on http://www.openser.org/pipermail/users/2006-January/002747.html seems to be very similar to mine. I am using openser-1.1.0-9etch1. I tried to replace the FROM variable as follows: loadmodule "uac.so" modparam("uac","from_restore_mode","auto") ... route { ... avp_write("$fU", "$avp(s:display)"); avp_write("$fU", "$avp(s:dispuri)"); avp_subst("$avp(s:display)", "/^36(.*)/06\1/"); avp_subst("$avp(s:dispuri)", "/^36(.*)/sip:06\1@datanet.hu/"); uac_replace_from("$avp(s:display)", "$avp(s:dispuri)"); ... } Unfortunately nothing seems to happen: no change is made on several VoIP telephones, including X-Lite as well. loose_route() and record_route() were also used in my configuration. Could you please help me what I am missing here? If needed, I can also send a full config file. Thank you in advance, Zoltan Kovacs sysadmin

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[Serusers] tm module problem
by Szasz Szabolcs 05 Dec '07

05 Dec '07

Hi I have a problem with tm module, I can't set the module parameter pass_provisional_replies . The module tm is loaded succesfully, I insert in to the configuration this line: modparam("tm", "pass_provisional_replies", 1) When I start ser I got this error message : 0(9618) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(9618) parse error (33,19-20): Can't set module parameter ser version used 0.9.6 , I couldn't found any solution on the google. I have to recompile the tm module with some modifications ? or I have other mistake in the configuration? Please help! Thank you Szasz Szabolcs

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[OpenSER-Users] Problem with setting the Acct-Authentic Radius Attribute
by Dan-Cristian Bogos 04 Dec '07

04 Dec '07

Folks, I am trying to make use of Acct-Authentic to identify authenticated sessions inside accounting packets. All ok, no problem with dictionaries, but when it comes to receiving the value in freeradius server, I get all sort of strange ones (integers). I am already using Acct-Authentic properly working with other software on the same radius server. Here is my config: modparam("acc", "radius_extra", " Called-Station-Id=$tu; Calling-Station-Id=$fu; Canonical-URI=$tu; User-Name=$au; Sip-User-Realm=$ar; Source-IP=$si; Source-Port=$sp; Acct-Authentic=$avp(s:acct_authentic); From-Header=$hdr(from); User-Agent=$hdr(user-agent); Contact=$hdr(contact); Event=$hdr(event)") I have tried different values for the $avp(s:acct_authentic): 1, "1", RADIUS, "RADIUS", etc. and each time I got a corresponding encoded. EG: If I will set the AVP through SIP-AVP or directly inside the routing script, for acct_authentic=1 I will receive in radius Acct-Authentic = 49. Are there any developers around who have implemented radius support in ser/openser and could help me troubleshooting this issue? My openser version: openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:3003M @(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $ Ta, DanB

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Re: [OpenSER-Users] OpenSER Sending INVITE twice
by Ovidiu Sas 04 Dec '07

04 Dec '07

There is no per upstream server config. Best thing to do is upgrade to openser 1.2. Starting with this release the timer granularity has been increased. If you have a high RTT, you can increase the T1 timer: http://www.openser.org/docs/modules/1.2.x/tm#AEN168 Hope this helps, Ovidiu Sas On Dec 4, 2007 12:44 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Oh, and RTT as OpenSER determines it to where? Is it one RTT for all > outgoing traffic or is it a different RTT for every upstream server? > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: Users(a)lists.openser.org > Sent: Monday, November 26, 2007 5:47:23 PM > Subject: Re: [OpenSER-Users] OpenSER Sending INVITE twice > > This is the way that is supposed to work. If nothing is received from > upstream, the request is resent. > From the rfc3261: > > 17.1.1.1 Overview of INVITE Transaction > > The INVITE transaction consists of a three-way handshake. The client > transaction sends an INVITE, the server transaction sends responses, > and the client transaction sends an ACK. For unreliable transports > (such as UDP), the client transaction retransmits requests at an > interval that starts at T1 seconds and doubles after every > retransmission. T1 is an estimate of the round-trip time (RTT), and > it defaults to 500 ms. > > > Regards, > Ovidiu Sas > > On Nov 26, 2007 3:29 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > > > I have an OpenSER 1.1 install here, and for some reason, OpenSER is > sending > > the INVITE message twice to the upstream host. > > > > The time difference between them is about 1/5 of a second. Nothing is > > received between the first and second INVITE's. > > > > Why is OpenSER doing this? > > > > I have xlog() statements everywhere in openser.cfg, and OpenSER is only > > logging ONE outgoing INVITE message, eventhough it's sending two. It is > > logging the multiple TRYING messages that come back however. > > > > OpenSER is not calling failure_route because nothing is logged in there. > > What could be going wrong? Why is it doing this? > > > > Doug. > > > > > > > > ________________________________ > > Get easy, one-click access to your favorites. Make Yahoo! your homepage. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now.

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[OpenSER-Users] invalid cseq for aor
by Gerson Matiolli 04 Dec '07

04 Dec '07

I used the SER for 2 years and not had this problem. I was using OPENSER for 20 days without problems. How can I fix this in Openser ? Thanks [OpenSER-Users] invalid cseq for aor Bogdan-Andrei Iancu bogdan at voice-system.ro Tue Dec 4 09:09:53 UTC 2007 * Previous message: [OpenSER-Users] invalid cseq for aor * Next message: [OpenSER-Users] perl * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi Gerson, The error is generated when a re-REGISTER (same contact and callid) is received but with invalid CSEQ number (equal or less than the previous). Regards, Bogdan Gerson Matiolli wrote: > Hi, All > > I am with the OPENSER running some time without problems. > > Yesterday, he went not to complete any call, and the system had the following message log: > > > ERROR:registry:update_contacts: invalid cseq for aor > ERROR:registry:update_contacts: invalid cseq for aor > ERROR:registry:update_contacts: invalid cseq for aor > > for all numbers > > Thanks >

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[OpenSER-Users] Remove header from Cancel
by Jarkko Nevala 04 Dec '07

04 Dec '07

Hello. Is it possible to remove Route: -header from Cancel requests. I have an user agent that inserts Route: -header in Cancel requests. I've already tried to get rid of it with remove_hf("Route") function, but it did not have any effect. From what I understand is, that this header is quite unnecessary in a stateful proxy, because the Cancel messages are handled hop-by-hop basis. It's just that it causes problems at the next hop proxy. I didn't even know about this problem with the UA before I started using the development version of OpenSER, the 1.2.2 version removed the Route: -header automatically before relaying Cancel forward. Thank you JN

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[OpenSER-Users] how does fix_nated_register() work?
by Daniel Grotti 04 Dec '07

04 Dec '07

Hi all, I have a user behind NAT with address : sip:userA@192.168.10.210:42276. When user send REGISTER, the proxy checks if user is behind NAT. If it is, proxy makes: fix_nated_register(); force_rport(); setbflag(6). When I check what kind of address proxy has saved on LOCATION database, I see at Contact column: sip:userA@IP_NAT:42276. So proxy mantains the same Port of the user and doesn't save the NAT's IP:PORT pair. Why? In this way, in fact, proxy is not able to keep connection alive. Regards, daniel

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[OpenSER-Users] Module Programming
by Daniel Corbe 04 Dec '07

04 Dec '07

Hello, Is there a better module programming guide than the one found on the docuwiki at http://www.openser.org/dokuwiki/doku.php/development:write-module ? It really doesn't explain much; for example the fact that functions are (or used to be) called with a paramater struct sip_msg *. The original SER 0.8.x and 0.9.x core had an extremely comprehensive guide to writing modules; however, I'm reluctant to use those texts as a guide for fear that the material is dated. Regards, Daniel

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[OpenSER-Users] Regarding CRBT
by Amit Vijayvargiya 04 Dec '07

04 Dec '07

Hi all, can anybody tell where we have to put the routing code in openser.cfg for CRBT(caller ring backtone ) application. A------------------------------------------make call -------------->B same time A------------------------------------------------------------------------->C in "C" we are playing prompts. Regards Amit

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[OpenSER-Users] Could Any body suggest me an SIP programming interface please?
by Samira Salam 04 Dec '07

04 Dec '07

Hi , As a part of my MS project I proposed a 3-party authentication protocol which can be used in SIP too. for SIP. The proxy server utilizes this protocol to authenticate the UC's Invite,using the Registrar server. Now, I have to estimate the totoal overhead of the scheme on a normal SIP call. I searched dozens of SIP, UMTS, GSM related papers to find a simulator or emulator which could be used to to achive the goal but I couldn't get any solutions as I need to consider execution of the RSA,AES and SHA on the agent's processor, I also need the solution to answer me with not too much programming due to my time constraints. I browsed the Resirocate project docs but I don't think then can answer me in time. I am also not sure whether SIPp, MS LIve communication server and SERcan help me? Could you please suggest me any rapid solutions? Any simulation techniques?any piece of software or programmining interface? By the way I am a little familiar with opnet as I did a tiny project with it. Doest it help me? Thanks in advance. Samira. --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

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[OpenSER-Users] openser: general question
by Helmut Kuper 04 Dec '07

04 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm quite unsure about openser's hangling of message processing. So can anybody answere following 2 questions please ? 1. When multiple messages come in, lets say 3 INVITEs. Will openser fork one child for each message, run the route script and terminate the child after that? Or will openser throw all 3 INVITEs at the same time into the route script? 2. If the 3 INVITEs share the same route script, are they handled sequentially? What happens to the AVPs and $var variables? Are they gonna mixed? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVRWA4tZeNddg3dwRAkQZAJ9Ruy3kr2RzODDlm55uZ4Y6yZTcPACgrl+v qmOW524gE6vgsWNgOalpfW4= =oR4D -----END PGP SIGNATURE-----

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[OpenSER-Users] Napting interval
by Daniel 04 Dec '07

04 Dec '07

Hi all, what is the different between napting interval done by nathelper-module and the other ones done by mediaproxy-module ? Can I set both of them at the same time ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Calls Between Registered Accounts
by VoIP Forums www.Go4Calls.com 04 Dec '07

04 Dec '07

Good day friends, I was trying to install Openser and Asterisk on same server. I need some configuration or information if anyone can point me. I need to forward only International call to the Asterisk but the same time if registered accounts of openser want to make call between them the can call each other without routing to asterisk. Could you please give us the right parameters of openser.cfg to route calls between openser accounts? Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[OpenSER-Users] invalid cseq for aor
by Gerson Matiolli 04 Dec '07

04 Dec '07

Hi, All I am with the OPENSER running some time without problems. Yesterday, he went not to complete any call, and the system had the following message log: ERROR:registry:update_contacts: invalid cseq for aor ERROR:registry:update_contacts: invalid cseq for aor ERROR:registry:update_contacts: invalid cseq for aor for all numbers Thanks

2 1

[OpenSER-Users] Help with Setup
by Arun Kumar 04 Dec '07

04 Dec '07

Hi, I need to configure my ser / openser in such a way that these sites should be able to call each other : Site A -> 01113, 01114 Site B -> 02113, 02114 Site C -> 03113, 03114 User in Site A (Dial 02113 or 03113) --------> SER / OpenSER --------> User in Site B or Site C and they should be able to call within site : User in Site A (Call 114) ----------> SER / OpenSER ---------> User in Site A here 01, 02, 03 are my site code so site code is only needed when users want to call between sites. If they are calling within same site no need to dial site id. please advise and give some links. Thanks, arun

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[Serusers] SER data model discussion - developers please read
by X Z 04 Dec '07

04 Dec '07

Hi All, This is specifically for the SER/OpenSER developers, but I'm not a serdev list member so I'm posting here. I've been using SER since version 0.8.X and I'm still running 0.8.14production for my company PBX to this day. I was very excited as version 2 became a release candidate and I downloaded it for testing. I was pretty disappointed with one aspect of the new data model and I'm requesting that the developers consider a further revision on the data model. Basically, taking all fields out of the subscriber table like Last_name, first_name, email, timezone, rpid/asserted identity, etc, etc is not the greatest idea. It's a better idea from a database architecture and performance perspective to keep adding columns into that table for data that has a 1 to 1 relationship with a user, and that is common in > 90% of SER's use cases (i.e. corporate, carrier/VSP.) I would suggest adding voicemail_password, and maybe every other field that is being added into the default attributes script that I saw in CVS recently. If you already know what attributes a user has (and they have a 1 to 1 relationship), then its far better from a db performance perspective to keep these attributes in the user table. I know that the code becomes more complicated, but I think it may be a tradeoff worth discussing. See this discussion ( http://asktom.oracle.com/pls/asktom/f?p=100:11:0::::P11_QUESTION_ID:1067808…) between Oracle users and Tom, (an Oracle engineer/architect.) The full text of this discussion is very informative and I highly recommend people read it through. Tom's conclusion is that the type of data model being discussed, and now being used in SER fails for all but the most trivial of applications. Maybe SER *by itself* qualifies as "trivial" from a database architect's perspective, but think about things like Asterisk integration, which is quite common. You quickly run into some very nasty queries . . . Please note that I am not a software developer nor a database engineer, just a user who reads a lot, so I'm open to being the ignorant one here, but I thought that this should be discussed among users and developers. Thanks for considering, Mahatma http://asktom.oracle.com/pls/asktom/f?p=100:11:0::::P11_QUESTION_ID:1067808… The following is an excerpt from the above link: Here is a excerpt from my forthcoming book where I talk about this (and show you how ugly, hard and inefficient queries against your very flexible model will be) (2)Do not use Generic Data Models Frequently I see applications built on a generic data model for "maximum flexibility" or applications built in ways that prohibit performance. Many times - these are one in the same thing! For example, it is well known you can represent any object in a database using just four tables: Create table objects ( oid int primary key, name varchar2(255) ); Create table attributes ( attrId int primary key, attrName varchar2(255), datatype varchar2(25) ); Create table object_Attributes ( oid int, attrId int, value varchar2(4000), primary key(oid,attrId) ); Create table Links ( oid1 int, oid2 int, primary key (oid1, oid2) ); That's it - no more CREATE TABLE for me! I can fill the attributes table up with rows like this: insert into attributes values ( 1, 'DATE_OF_BIRTH', 'DATE' ); insert into attributes values ( 2, 'FIRST_NAME', 'STRING' ); insert into attributes values ( 3, 'LAST_NAME', 'STRING' ); commit; And now I'm ready to create a PERSON record: insert into objects values ( 1, 'PERSON' ); insert into object_Attributes values( 1, 1, '15-mar-1965' ); insert into object_Attributes values( 1, 2, 'Thomas' ); insert into object_Attributes values( 1, 3, 'Kyte' ); commit; insert into objects values ( 2, 'PERSON' ); insert into object_Attributes values( 2, 1, '21-oct-1968' ); insert into object_Attributes values( 2, 2, 'John' ); insert into object_Attributes values( 2, 3, 'Smith' ); commit; And since I'm good at SQL, I can even query this record up to get the FIRST_NAME and LAST_NAME of all PERSON records: ops$tkyte@ORA920> select max( decode(attrName, 'FIRST_NAME', value, null )) first_name, 2 max( decode( attrName, 'LAST_NAME', value, null ) ) last_name 3 from objects, object_attributes, attributes 4 where attributes.attrName in ( 'FIRST_NAME', 'LAST_NAME' ) 5 and object_attributes.attrId = attributes.attrId 6 and object_attributes.oid = objects.oid 7 and objects.name = 'PERSON' 8 group by objects.oid 9 / FIRST_NAME LAST_NAME -------------------- -------------------- Thomas Kyte John Smith Looks great, right? I mean, the developers don't have to create tables anymore, we can add columns at the drop of a hat (just requires an insert into the ATTRIBUTES table). The developers can do whatever they want and the DBA can't stop them. This is ultimate "flexibility". I've seen people try to build entire systems on this model. But, how does it perform? Miserably, terribly, horribly. A simple "select first_name, last_name from person" query is transformed into a 3-table join with aggregates and all. Further, if the attributes are "NULLABLE" - that is, there might not be a row in OBJECT_ATTRIBUTES for some attributes, you may have to outer join instead of just joining which in some cases can remove more optimal query plans from consideration. Writing queries might look pretty straightforward, but it's impossible to do in a performant fashion. For example, if we wanted to get everyone that was born in MARCH or has a LAST_NAME = 'SMITH', we could simply take the query from above and just wrap an inline view around that: ops$tkyte@ORA920> select * 2 from ( 3 select max(decode(attrName, 'FIRST_NAME', value, null)) first_name, 4 max(decode(attrName, 'LAST_NAME', value, null)) last_name, 5 max(decode(attrName, 'DATE_OF_BIRTH', value, null)) date_of_birth 6 from objects, object_attributes, attributes 7 where attributes.attrName in ( 'FIRST_NAME', 'LAST_NAME', 'DATE_OF_BIRTH' ) 8 and object_attributes.attrId = attributes.attrId 9 and object_attributes.oid = objects.oid 10 and objects.name = 'PERSON' 11 group by objects.oid 12 ) 13 where last_name = 'Smith' 14 or date_of_birth like '%-mar-%' 15 / FIRST_NAME LAST_NAME DATE_OF_BIRTH -------------------- -------------------- -------------------- Thomas Kyte 15-mar-1965 John Smith 21-oct-1968 So, it looks "easy" to query, but think about the performance! If you had a couple thousand OBJECT records, and a couple tens of thousands of OBJECT_ATTRIBUTES - Oracle would have to process the entire inner group by query first and then apply the WHERE clause. This is not a made up data model, one that I crafted just to make a point. This is an actual data model that I've seen people try to use. Their goal is ultimate flexibility. They don't know what OBJECTS they need, they don't know what ATTRIBUTES they will have. Well - that is what the database was written for in the first place: Oracle implemented this thing called SQL to define OBJECTS and ATTRIBUTES and lets you use SQL to query them. You are trying to put a generic layer on top of a generic layer - and it fails each and every time except for the most trivial of applications.

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Re: [OpenSER-Users] OpenSER Sending INVITE Twice!?
by Ovidiu Sas 03 Dec '07

03 Dec '07

Hi Douglas, You will need to upgrade to 1.2.2. This issue was fixed starting with 1.2.0. Regards, Ovidiu Sas On Dec 3, 2007 3:12 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Ok, I can accept that it might be a timeout since it doesn't happen every > single time. > 1/5s seems awefully short though. How can I increase this? I can't find > anything in the docs about increasing this timeout... > > Doug. > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Tavis P. <tavis.lists(a)galaxytelecom.net> > Cc: users(a)openser.org > Sent: Monday, December 3, 2007 12:01:53 PM > Subject: Re: [OpenSER-Users] OpenSER Sending INVITE Twice!? > > Right, 1/5 is not 1/2 :) > > -ovi > > On Dec 3, 2007 10:10 AM, Tavis P. <tavis.lists(a)galaxytelecom.net> wrote: > > > > It may also be related to the timer accuracy of OpenSER 1.1 > > > > If the T1 interval is 500ms but OpenSER 1.1 doesn't support sub-second > > timer accuracy it is probable to see it retransmitting in situations where > > T1 hasn't been reached yet > > It shouldn't cause any problems though, just a few erroneous retransmits > > from time to time > > > > tavis > > > > Ovidiu Sas wrote: > > Hi Douglas, > > > > You posted this questionawhileago and by checkingthe archives > > Irealised that my answer was truncated somehow ... > > Here's the full answer: > > > > This is the way that is supposed to work. If nothing is received from > > upstream, the request is resent. > > >From the rfc3261: > > > > 17.1.1.1 Overview of INVITE Transaction > > > > The INVITE transaction consists of a three-way handshake. The client > > transaction sends an INVITE, the server transaction sends responses, > > and the client transaction sends an ACK. For unreliable transports > > (such as UDP), the client transaction retransmits requests at an > > interval that starts at T1 seconds and doubles after every > > retransmission. T1 is an estimate of the round-trip time (RTT), and > > it defaults to 500 ms. > > > > > > It seems that your upstream UAS is slowin sending out the 100 Trying > > and therefor openSER is resending out the INVITE. > > > > > > Regards, > > Ovidiu Sas > > > > On Dec 3, 2007 12:35 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > > > > > I have an OpenSER 1.1 install here, and for some reason, OpenSER is > sending > > the INVITE message twice to the upstream host. > > > > Asterisk OpenSER Provider > > ----------------------------------------------- > > | ---INVITE---> | | > > | | ---INVITE---> | > > | | ---INVITE---> | > > | | <---TRYING--- | > > | | <---TRYING--- | > > > > The time difference between them is about 1/5 of a second. Nothing is > > received between the first and second INVITE's. > > > > Why is OpenSER doing this? > > > > I have xlog() statements everywhere in openser.cfg, and OpenSER is only > > logging ONE outgoing INVITE message, eventhough it's sending two. It is > > logging the multiple TRYING messages that come back however. > > > > OpenSER is not calling failure_route because nothing is logged in there. > > What could be going wrong? Why is it doing this? > > > > Doug. > > > > ________________________________ > > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > > now. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > ________________________________ > Get easy, one-click access to your favorites. Make Yahoo! your homepage.

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[OpenSER-Users] OpenSER Sending INVITE Twice!?
by Douglas Garstang 03 Dec '07

03 Dec '07

I have an OpenSER 1.1 install here, and for some reason, OpenSER is sending the INVITE message twice to the upstream host. Asterisk OpenSER Provider ----------------------------------------------- | ---INVITE---> | | | | ---INVITE---> | | | ---INVITE---> | | | <---TRYING--- | | | <---TRYING--- | The time difference between them is about 1/5 of a second. Nothing is received between the first and second INVITE's. Why is OpenSER doing this? I have xlog() statements everywhere in openser.cfg, and OpenSER is only logging ONE outgoing INVITE message, eventhough it's sending two. It is logging the multiple TRYING messages that come back however. OpenSER is not calling failure_route because nothing is logged in there. What could be going wrong? Why is it doing this? Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] Multidomain Support
by Arun Kumar 03 Dec '07

03 Dec '07

Hi, I'm going to setup multi domain ser or openser , if some one can have any sample configuration of some good guide to configure. Please help. thanks Arun

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[OpenSER-Users] why mediaproxy doesn't works?
by Daniel Grotti 03 Dec '07

03 Dec '07

Hi all, I'm usign 2 users behind the same Nat and I'm using x-lite 3.0. User A calls user B. I want to use mediaproxy between 2 users and I have configured my openser file in that way. Mediaproxy doesn't works, and I've noticed diferences between the "200 OK" message received by user A and that sent by proxy. user A --- INVITE ----> proxy --- INVITE-----> user B user A <----200 OK ---- proxy < ----- 200 OK --- user B this is the 200 OK message sent by proxy to USER A. This has been already manipulated by proxy (chenge IP and PORT in SDP part) for using mediaproxy : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.245:14620;received=85.18.14.25;branch=z9hG4bK-d87543-37310b5f2633e814-1--d87543-;rport=18267 Record-Route: <sip:137.204.107.138:5060;lr;nat=yes;ftag=910e1011> Contact: <sip:daniel@85.18.14.25:20881;rinstance=8a9df9cdfc3da728> To: "daniel"<sip:daniel@137.204.107.138>;tag=6e33c440 From: "daniel.grotti"<sip:daniel.grotti@137.204.107.138>;tag=910e1011 Call-ID: Y2M3ZDNjOGExN2Q0YTBlYjViMTg1NDAwOTAxZjliMDY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: *375* v=0 o=- 2 2 IN IP4 192.168.10.224 s=CounterPath X-Lite 3.0 *c=IN IP4 137.204.107.138 IP of media relay* t=0 0 *m=audio 60004 RTP/AVP 107 119 100 106 0 105 98 8 101 # 60004 port of mediarelay* a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv and this is the message sniffed by client A: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.245:14620;received=85.18.14.25;branch=z9hG4bK-d87543-37310b5f2633e814-1--d87543-;rport=18267 Record-Route: <sip:137.204.107.138:5060;lr;nat=yes;ftag=910e1011> Contact: <sip:daniel@85.18.14.25:20881;rinstance=8a9df9cdfc3da728> To: "daniel"<sip:daniel@137.204.107.138>;tag=6e33c440 From: "daniel.grotti"<sip:daniel.grotti@137.204.107.138>;tag=910e1011 Call-ID: Y2M3ZDNjOGExN2Q0YTBlYjViMTg1NDAwOTAxZjliMDY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: *374* v=0 o=- 2 2 IN IP4 192.168.10.224 s=CounterPath X-Lite 3.0 *c=IN IP4 192.168.10.224* t=0 0 *m=audio 36124 RTP/AVP 107 119 100 106 0 105 98 8 101* a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv It's the same message but with c and m parameter changed. So, mediaproxy doesn't work and 2 users are directly connected. any suggestion? regards, daniel

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Re: [OpenSER-Users] [Serusers] Multidomain Support
by Jiri Kuthan 03 Dec '07

03 Dec '07

Default SER configuration does that, you just need command-line tools to add the domain. -jiri At 16:41 03/12/2007, Arun Kumar wrote: >Hi, > >I'm going to setup multi domain ser or openser , if some one can have any sample configuration of some good guide to configure. Please help. > > >thanks > >Arun >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [OpenSER-Users] perl
by David Villasmil Govea 03 Dec '07

03 Dec '07

> > > Message: 9 > Date: Mon, 3 Dec 2007 15:59:06 +0100 > From: Bastian Friedrich <bastian.friedrich(a)collax.com> > Subject: Re: [OpenSER-Users] perl > To: users(a)lists.openser.org > Message-ID: <200712031559.06986.bastian.friedrich(a)collax.com> > Content-Type: text/plain; charset="iso-8859-15" > > On Monday 03 December 2007, David Villasmil Govea wrote: > [...] > > 0(0) perl error: Can't locate OpenSER/Utils/PhoneNumber.pm in @INC > [...] > > > > Although modules are there in: > > > > No, there is no "PhoneNumber.pm" in that path. > > [...] > > me@server:/usr/local/lib/openser/perl# ls -l > > /usr/local/lib/openser/perl/OpenSER/Utils/ > > total 8 > > -rw-r--r-- 1 root root 6096 2007-12-03 14:00 PhoneNumbers.pm > > It's "PhoneNumbers.pm". Correct typing is _not_ optional while programming > :) thanks for the heads-up.. but i didn't type a thing, its all in the tar file... so there's a typo there. so THIS should be the correct /usr/local/etc/openser/openser.pl file, which comes with the tar ball: ************************************************************************************************************** push @INC, '/usr/local/lib/openser/perl/'; use OpenSER::Utils::PhoneNumbers; OpenSER->import; my $phonenumber = new OpenSER::Utils::PhoneNumbers( publicAccessPrefix => "0", internationalPrefix => "+", longDistancePrefix => "0", areaCode => "761", pbxCode => "456842", countryCode => "49" ); $canonical = canonicalForm("07612034567"); $number = dialNumber("+497612034567"); exit 0; ************************************************************************************************************** thanks > > > > > > - Is there a tutorial for using MySQL as a backend? How about a > > > > cluster? > > > > > > MySQL backend for what? > > > > I have it working as a MySQL backend for users. What i need is the > logging > > of cdrs... i understand that there are several records for each call. > But, > > how do i configure this? is it possible with MySQL? thanks a lot. > > You should start a different thread with a different subject. > > What you are looking for seems to be accounting, provided by the OpenSER > module "acc". Module documentation is available on > http://www.openser.org/docs/modules/1.3.x/acc.html > The "acc" module works fine with MySQL. > > Bastian > > -- > Collax GmbH . Burkheimer Stra?e 3 . 79111 Freiburg . Germany > p: +49 (0) 761-45684-24 > f: +49 (0) 761-45684-10 www.collax.com > > Gesch?ftsf?hrer: William K. Hite / Boris Nalbach > AG M?nchen HRB 158898 . Ust.-IdNr: DE 814464942 > \ There are 2 ways to handle women and I know neither. > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 31, Issue 5 > ************************************ > -- DVG -- Imagination is more important than knowledge Albert Einstein

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[OpenSER-Users] openser trunk: presence bug?
by Helmut Kuper 03 Dec '07

03 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I use openser 1.3 latest trunk and try to use presence module. If presence server receives PUBLISH message the following error occurs on console: Dec 3 13:24:03 [616] ERROR:presence:handle_publish: Missing or unsupported event header field value Dec 3 13:24:03 [616] ERROR:presence:handle_publish: event=[presence] My openser config is this: xlog("Handle $rm\n"); # presence handling if(is_method("PUBLISH")) { handle_publish(); xlog("DONE!\n"); sl_send_reply("200","PUBLISH ok "); t_release(); } Is this a bug or do I have to define the events somewhere? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHU/cP4tZeNddg3dwRAujxAKCEG2zdB7nhqychuMcgf6oEurky4gCdHH/v NGJILGEwcG7Xc3mwfQytKFY= =Wwcp -----END PGP SIGNATURE-----

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[OpenSER-Users] perl
by David Villasmil Govea 03 Dec '07

03 Dec '07

> > > OpenSER modules can be included while compiling by using the > "include_modules" > statement. Thus, the perl module (not "mod_perl", as Daniel already > pointed > out) can be compiled along with OpenSER by using > make include_modules="perl" > > You will need your distribution's perl development packages installed > while > compiling. Yes, I have it compiled with no problems, but when i try to load a perl script, i get this: 0(0) perl: Setting lib path: '/usr/local/lib/openser/perl/' 0(0) perl error: Can't locate OpenSER/Utils/PhoneNumber.pm in @INC (@INC contains: /usr/local/lib/openser/perl/ /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at /usr/local/etc/openser/openser.pl line 4. 0(0) perl error: Can't locate OpenSER/Utils/PhoneNumber.pm in @INC (@INC contains: /usr/local/lib/openser/perl/ /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at /usr/local/etc/openser/openser.pl line 4. BEGIN failed--compilation aborted at /usr/local/etc/openser/openser.pl line 4. Can't locate OpenSER/Utils/PhoneNumber.pm in @INC (@INC contains: /usr/local/lib/openser/perl/ /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at /usr/local/etc/openser/openser.pl line 4. BEGIN failed--compilation aborted at /usr/local/etc/openser/openser.pl line 4. Although modules are there in: me@server:/usr/local/lib/openser/perl# ls -l /usr/local/lib/openser/perl/ total 4 drwxrwx--- 4 root root 168 2007-12-03 14:00 OpenSER -rwxrwx--- 1 root root 1354 2007-12-03 14:00 OpenSER.pm me@server:/usr/local/lib/openser/perl# ls -l /usr/local/lib/openser/perl/OpenSER total 12 -rw-r--r-- 1 root root 4844 2007-12-03 14:00 Constants.pm drwxr-xr-x 2 root root 120 2007-12-03 14:00 LDAPUtils -rw-r--r-- 1 root root 1602 2007-12-03 14:00 Message.pm drwxr-xr-x 2 root root 80 2007-12-03 14:00 Utils me@server:/usr/local/lib/openser/perl# ls -l /usr/local/lib/openser/perl/OpenSER/Utils/ total 8 -rw-r--r-- 1 root root 6096 2007-12-03 14:00 PhoneNumbers.pm > > > Feel free to ask if you have any concrete questions. > > > - Is there a tutorial for using MySQL as a backend? How about a > > cluster? > > MySQL backend for what? I have it working as a MySQL backend for users. What i need is the logging of cdrs... i understand that there are several records for each call. But, how do i configure this? is it possible with MySQL? thanks a lot. > > > Bastian > thanks much for your help! David -- DVG -- Imagination is more important than knowledge Albert Einstein

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[Serusers] Problems with syslog and mysql on version 0.96
by Michael Roeper 03 Dec '07

03 Dec '07

Hi, we have SER version 0.96 running on a debian system using mysql for user authentication. Everything works fine as long as we use the standard logging to /var/log/syslog. When we change the logging to use another syslog facility (e.g. log_facility=LOG_LOCAL0 in ser.cfg) all messages are written to the file specified in syslog.conf, but the users are not able to register to ser. It looks as if the db-connection to the mysql database is not available. Has anybody observed this behaviour and can provide a hint ? We appreciate any kind of help. Kind regards, Michael -- Dr.Michael Roeper Email: m.roeper(a)wenzel-elektronik.de Wenzel Elektronik GmbH Tel.: 04121/4872-69 Max-Planck-Str. 2-4 Fax : 04121/4872-18 25335 Elmshorn http://www.wenzel-elektronik.de Sitz der Gesellschaft: Elmshorn Registergericht: Pinneberg, HRB 1974 EL USt-IdNr.: DE 812946778 Geschäftsführung: Dipl. Ing Jörg Simon, Dr. Volker Henkel --

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[OpenSER-Users] OpenSER: t-check_trans question
by Helmut Kuper 03 Dec '07

03 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, today I found, that t_check_trans function of TM module seems to has a problem with at least PUBLISH and SUBSCRIBE Messages. If openser checks theses messages with t_check_trans() route script execution will stop. Is this intended or a bug? Module docc says, that t_check_trans() cares about ACKs and CANCELs .. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHU9mU4tZeNddg3dwRAlk/AJ9k/B/YBum72U45csWBqyi7RMLxXQCfWLLx rxRkJmYXFSOFvq6FlYKR3ng= =0bIH -----END PGP SIGNATURE-----

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[Serusers] function get_from_did(&did, msg) error
by Tomasz Zieleniewski 03 Dec '07

03 Dec '07

Hi, I was trying to set up media proxy module and after some time of debuging I discovered that the problems were in the get_from_did(&did, msg) function. in mediaproxy.c file in line 469 there is such piece of code: if (get_from_did(&did, msg) == 0) { *type = "local"; return did; } when this code is invoked the result is wrong. this is some nowhere pointer with the length of the field unpredictable. after this invocation there is another piece of code wich tries to allocate some memory baseing on the length of the did string.(line 1449) cmdlen = callId.len + strlen(clientIP) + fromDomain.len + toDomain.len + userAgent.len*3 + infolen + 128; command = pkg_malloc(cmdlen); if (!command) { LOG(L_ERR, "error: use_media_proxy() - (!command): out of memory\n"); return -1; } this causes out of memory error when value of fromDomain.len gives random big number(below example of one my shots) Dec 2 18:17:17 rd ser[5884]: route[MEDIA]: use_media_proxy() Dec 2 18:17:17 rd ser[5884]: error: get_from_did - true Dec 2 18:17:17 rd ser[5884]: error: callId.len: 44 Dec 2 18:17:17 rd ser[5884]: error: strlen(clientIP): 14 Dec 2 18:17:17 rd ser[5884]: error: fromDomain.len: 1235653920 Dec 2 18:17:17 rd ser[5884]: error: toDomain.len: 14 Dec 2 18:17:17 rd ser[5884]: error: userAgent.len*3: 96 Dec 2 18:17:17 rd ser[5884]: error: infolen: 107 Dec 2 18:17:17 rd ser[5884]: error: cmdlen 1235654349 Dec 2 18:17:17 rd ser[5884]: error: use_media_proxy() - (!command): out of memory please, can someone check this and fix best regards tomasz

2 2

[OpenSER-Users] mod_perl
by David Villasmil Govea 03 Dec '07

03 Dec '07

Hello all, Its my first time writing to the list, i hope someone can help me out. - Can anyone point me in the right direction for compiling with mod_perl? I tried compiling but it wouldn't work. That was a few months ago, though. Is there a good step-by-step tutorial? - Is there a tutorial for using MySQL as a backend? How about a cluster? thanks to all. David

3 2

[Serusers] error in mediaproxy - module generates wrong request format
by Tomasz Zieleniewski 02 Dec '07

02 Dec '07

Hi, I am using mediaproxy for rtp traffic. I am using ser rel_2_0_0_rc5 and media proxy 1.9.0. When there is an invite message and i invoke use_media_proxy media proxy module in ser send wrong request to proxydispatcher or mediaproxy (depending on the type of configuration) The lookup request format according to the modules/request.py file is 'cmd', 'sessionId', 'streams', 'signalingIP', 'callerDomain', 'callerType','calledDomain', 'calledType', 'userAgent', 'info' but mediaproxy module sends such one: lookup MjI3MjcwMzA3NGQxNTc3YjUxZDA5NmQ1NzE4ZmNiOTQ. 192.168.129.74:13122:audio 192.168.129.74 local voip.touk.pl remote Asterisk=20PBX info= from:tzl@voip.touk.pl,to:poczta@voip.touk.pl ,fromtag:cb239428,totag:as707ae074 there is no callerDomain attribute. Please tell me did I miss something or this is realy bug in mediaproxy module. Cheers -tomasz

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Re: [OpenSER-Users] Users Digest, Vol 31, Issue 2
by David Villasmil Govea 02 Dec '07

02 Dec '07

Oops! sorry, wrong list!

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[OpenSER-Users] compiling woomera for FS
by David Villasmil Govea 02 Dec '07

02 Dec '07

Hello all again, I jus compiled beta2 of FS perfectly. I want to use woomera H.323 but it is simply impossible to do. I tried to follow the instructions on the wiki but it simply doesn't work. pwlib and openh323 compiled fine but woomera comes back with: g++ -D_REENTRANT -Wall -I/root/openh323/../pwlib/include -DPTRACING -I/root/openh323/include -Os -felide-constructors -Wreorder -c main.cxx -o obj_linux_x86_r/main.o In file included from main.cxx:129: openh323.h:212:22: warning: no newline at end of file /root/openh323/../pwlib/include/ptclib/pwavfile.h:166: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:172: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::ChunkHeaderâ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:173: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:178: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::ChunkHeaderâ?T lib/woomera.h:819: warning: â?~class Woomera::ExternalCommandâ?T has virtual functions but non-virtual destructor /root/openh323/include/h4601.h: In member function â?~H460_FeatureContent::operator H460_FeatureTable*()â?T: /root/openh323/include/h4601.h:292: warning: type-punning to incomplete type might break strict-aliasing rules after repeating the lines above several times: g++ -D_REENTRANT -Wall -I/root/openh323/../pwlib/include -DPTRACING -I/root/openh323/include -Os -felide-constructors -Wr eorder -c wivr.cxx -o obj_linux_x86_r/wivr.o /root/openh323/../pwlib/include/ptclib/pwavfile.h:166: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:172: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::Chu nkHeaderâ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:173: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:178: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::Chu nkHeaderâ?T ./lib/woomera.h:819: warning: â?~class Woomera::ExternalCommandâ?T has virtual functions but non-virtual destructor g++ -o obj_linux_x86_r/woomera -L/root/openh323/../pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o ./obj_linux_ x86_r/openh323.o ./obj_linux_x86_r/woomera.o ./obj_linux_x86_r/server.o ./obj_linux_x86_r/connection.o ./obj_linux_x86_r/ client.o ./obj_linux_x86_r/wivr.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lresolv -ldl ./obj_linux_x86_r/openh323.o: In function `OpenH323WoomeraCall::OnH323OpenAudioChannel(H323AudioCodec&, int)': openh323.cxx:(.text+0x2b49): undefined reference to `PVXMLSession::PVXMLSession(PTextToSpeech*, int)' openh323.cxx:(.text+0x2ba2): undefined reference to `PVXMLSession::SetTextToSpeech(PString const&)' ./obj_linux_x86_r/woomera.o: In function `WoomeraPlayableToneDevice::Open(PVXMLChannel&, PString const&, int, int, int)': woomera.cxx:(.text+0x1239): undefined reference to `PVXMLPlayableData::SetData(PBYTEArray const&)' woomera.cxx:(.text+0x1253): undefined reference to `PVXMLPlayableData::Open(PVXMLChannel&, PString const&, int, int, int)' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayable::~PVXMLPlayable()': woomera.cxx:(.text._ZN13PVXMLPlayableD2Ev[PVXMLPlayable::~PVXMLPlayable()]+0x10): undefined reference to `vtable for PVXMLPla yable' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableData::~PVXMLPlayableData()': woomera.cxx:(.text._ZN17PVXMLPlayableDataD2Ev[PVXMLPlayableData::~PVXMLPlayableData()]+0x10): undefined reference to `vtable for PVXMLPlayableData' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableFilename::~PVXMLPlayableFilename()': woomera.cxx:(.text._ZN21PVXMLPlayableFilenameD2Ev[PVXMLPlayableFilename::~PVXMLPlayableFilename()]+0x10): undefined reference to `vtable for PVXMLPlayableFilename' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableURL::~PVXMLPlayableURL()': woomera.cxx:(.text._ZN16PVXMLPlayableURLD2Ev[PVXMLPlayableURL::~PVXMLPlayableURL()]+0x10): undefined reference to `vtable for PVXMLPlayableURL' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayable::PVXMLPlayable()': woomera.cxx:(.text._ZN13PVXMLPlayableC2Ev[PVXMLPlayable::PVXMLPlayable()]+0xe): undefined reference to `vtable for PVXMLPlaya ble' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableData::PVXMLPlayableData()': woomera.cxx:(.text._ZN17PVXMLPlayableDataC2Ev[PVXMLPlayableData::PVXMLPlayableData()]+0x13): undefined reference to `vtable f or PVXMLPlayableData' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableFilename::PVXMLPlayableFilename()': woomera.cxx:(.text._ZN21PVXMLPlayableFilenameC2Ev[PVXMLPlayableFilename::PVXMLPlayableFilename()]+0x16): undefined reference to `vtable for PVXMLPlayableFilename' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableURL::PVXMLPlayableURL()': woomera.cxx:(.text._ZN16PVXMLPlayableURLC2Ev[PVXMLPlayableURL::PVXMLPlayableURL()]+0x16): undefined reference to `vtable for PVXMLPlayableURL' ./obj_linux_x86_r/woomera.o: In function `WoomeraPlayableFilename::OnStop()': woomera.cxx:(.text._ZN23WoomeraPlayableFilename6OnStopEv[WoomeraPlayableFilename::OnStop()]+0xf): undefined reference to `PVX MLPlayableFilename::OnStop()' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV25WoomeraPlayableToneDevice[vtable for WoomeraPlayableToneDevice]+0x38): undefined r eference to `PVXMLPlayableData::Play(PVXMLChannel&)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV25WoomeraPlayableToneDevice[vtable for WoomeraPlayableToneDevice]+0x54): undefined r eference to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV25WoomeraPlayableToneDevice[vtable for WoomeraPlayableToneDevice]+0x58): undefined r eference to `PVXMLPlayableData::Rewind(PChannel*)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV26WoomeraPlayableSoundDevice[vtable for WoomeraPlayableSoundDevice]+0x54): undefined reference to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI25WoomeraPlayableToneDevice[typeinfo for WoomeraPlayableToneDevice]+0x10): undefined reference to `typeinfo for PVXMLPlayableData' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI26WoomeraPlayableSoundDevice[typeinfo for WoomeraPlayableSoundDevice]+0x10): undefin ed reference to `typeinfo for PVXMLPlayable' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x34): undefined refer ence to `PVXMLPlayableFilename::Open(PVXMLChannel&, PString const&, int, int, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x38): undefined refer ence to `PVXMLPlayableFilename::Play(PVXMLChannel&)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x54): undefined refer ence to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x58): undefined refer ence to `PVXMLPlayableFilename::Rewind(PChannel*)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI23WoomeraPlayableFilename[typeinfo for WoomeraPlayableFilename]+0x10): undefined ref erence to `typeinfo for PVXMLPlayableFilename' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV18WoomeraPlayableURL[vtable for WoomeraPlayableURL]+0x34): undefined reference to `P VXMLPlayableURL::Open(PVXMLChannel&, PString const&, int, int, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV18WoomeraPlayableURL[vtable for WoomeraPlayableURL]+0x38): undefined reference to `P VXMLPlayableURL::Play(PVXMLChannel&)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV18WoomeraPlayableURL[vtable for WoomeraPlayableURL]+0x54): undefined reference to `P VXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI18WoomeraPlayableURL[typeinfo for WoomeraPlayableURL]+0x10): undefined reference to `typeinfo for PVXMLPlayableURL' ./obj_linux_x86_r/server.o: In function `Woomera::Call::SetVar(PString const&, PString const&)': server.cxx:(.text+0x59c): undefined reference to `PVXMLSession::SetTextToSpeech(PString const&)' ./obj_linux_x86_r/server.o: In function `Woomera::Call::OnCmdRecord(PStringArray const&, PMIMEInfo const&, PString const&)': server.cxx:(.text+0x70bc): undefined reference to `PVXMLChannel::EndRecording()' ./obj_linux_x86_r/server.o: In function `Woomera::Call::OnCmdSpeak(PStringArray const&, PMIMEInfo const&, PString const&)': server.cxx:(.text+0x7e79): undefined reference to `PVXMLSession::ConvertTextToFilenameList(PString const&, PTextToSpeech::Tex tType, PStringArray&, int)' server.cxx:(.text+0x8053): undefined reference to `PVXMLPlayableFilenameList::Open(PVXMLChannel&, PStringArray const&, int, i nt, int)' ./obj_linux_x86_r/server.o: In function `PVXMLPlayableFilenameList::~PVXMLPlayableFilenameList()': server.cxx:(.text._ZN25PVXMLPlayableFilenameListD2Ev[PVXMLPlayableFilenameList::~PVXMLPlayableFilenameList()]+0x10): undefine d reference to `vtable for PVXMLPlayableFilenameList' ./obj_linux_x86_r/server.o: In function `PVXMLRecordableFilename::~PVXMLRecordableFilename()': server.cxx:(.text._ZN23PVXMLRecordableFilenameD2Ev[PVXMLRecordableFilename::~PVXMLRecordableFilename()]+0x10): undefined refe rence to `vtable for PVXMLRecordableFilename' ./obj_linux_x86_r/server.o: In function `PVXMLPlayableFilenameList::PVXMLPlayableFilenameList()': server.cxx:(.text._ZN25PVXMLPlayableFilenameListC2Ev[PVXMLPlayableFilenameList::PVXMLPlayableFilenameList()]+0x13): undefined reference to `vtable for PVXMLPlayableFilenameList' ./obj_linux_x86_r/server.o: In function `WoomeraPlayableFilenameList::OnStop()': server.cxx:(.text._ZN27WoomeraPlayableFilenameList6OnStopEv[WoomeraPlayableFilenameList::OnStop()]+0xf): undefined reference to `PVXMLPlayableFilenameList::OnStop()' ./obj_linux_x86_r/server.o: In function `PVXMLRecordableFilename::PVXMLRecordableFilename()': server.cxx:(.text._ZN23PVXMLRecordableFilenameC2Ev[PVXMLRecordableFilename::PVXMLRecordableFilename()]+0x16): undefined refer ence to `vtable for PVXMLRecordableFilename' ./obj_linux_x86_r/server.o:(.rodata._ZTV27WoomeraPlayableFilenameList[vtable for WoomeraPlayableFilenameList]+0x3c): undefine d reference to `PVXMLPlayableFilenameList::OnRepeat(PVXMLChannel&)' ./obj_linux_x86_r/server.o:(.rodata._ZTV27WoomeraPlayableFilenameList[vtable for WoomeraPlayableFilenameList]+0x54): undefine d reference to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/server.o:(.rodata._ZTI27WoomeraPlayableFilenameList[typeinfo for WoomeraPlayableFilenameList]+0x10): undefi ned reference to `typeinfo for PVXMLPlayableFilenameList' ./obj_linux_x86_r/server.o:(.rodata._ZTV25WoomeraRecordableFilename[vtable for WoomeraRecordableFilename]+0x30): undefined re ference to `PVXMLRecordableFilename::Open(PString const&)' . . . . . wivr.cxx:(.text+0x5464): undefined reference to `PXMLElement::GetElement(int) const' wivr.cxx:(.text+0x5529): undefined reference to `PXML::~PXML()' wivr.cxx:(.text+0x553a): undefined reference to `PXML::~PXML()' collect2: ld returned 1 exit status make: *** [obj_linux_x86_r/woomera] Error 1 root@icetech-server:~/woomera# What's the problem? Any help? Thanks a lot David

1 0

[OpenSER-Users] Replace Openser dns based sip location service by a DHT
by Victor Pascual Ávila 01 Dec '07

01 Dec '07

Hi all, we are working on a P2PSIP prototype and we are considering to couple a DHT Chord-based node with Openser. Current dns based sip location service would be replaced by a DHT, allowing several proxies/registrars to read/write overlays resource records. Any hint or guidelines would be appreciated. Thanks in advance and sorry for cross posting, I'm not sure this issue is related to devel or users. -- Victor Pascual Ávila

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[OpenSER-Users] Changing From: and To: URIs...
by Phil D'Amore 30 Nov '07

30 Nov '07

Hi Everyone... I'm looking for some advice. Let's say I have a To: header in an initial INVITE: To: Bob <sip:8005551212@sip.example.com> Due to carrier requirements, it really needs to be: To: Bob <sip:+18005551212@sip.example.com> I'm not asking how to do it. I know I can use some textops functions to make this happen. I also know that doing this might break something. The uac module already nicely handles the From: header in a way that doesn't break things. I thought about extending this to handle To:, but I worried that the Record-Route header might get too long for some UAs if I wind up having to change both headers. The To tag must also be dealt with somehow, which I think might make it harder to do than From. Using a 302 response is not an option for me, and I'm really trying to avoid getting a B2BUA involved with this right now. Moving from openser 1.1.1 to 1.3 is enough for me without changing my architecture ;). I'm wondering what would be the best way given what openser can do right now (1.3) to re-write that header that doesn't break anything. I'm not adverse to writing some code to make it happen if that is what is needed. Any ideas? Thanks, Phil

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