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37454 discussions

[Serusers] call via several SER server
by 静于 07 Aug '08

07 Aug '08

hello all, Could two UAs call each other via several SER proxy? I want to implement this function.How to do it?Do I need to modify the ser.cfg? How dose SER server connect each other? --------------------------------- 雅虎邮箱，您的终生邮箱！

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[Kamailio-Users] Problem using additional attributes through radius_extra parameter
by Ruchir 06 Aug '08

06 Aug '08

Hi, I'm trying to configure openser & freeradius with AAA. I'm using custom mysql tables for authentication & accounting. I've modified queries in sql.conf in raddb folder. Authentication works fine. I've added additional fields in radius_extra parameter of acc module. I've added those parameters in dictionaries of radiusclient-ng & freeradius. But I get blank values for custom fields. Any idea what can be wrong? *openser.cfg* modparam("acc", "radius_extra","User-Name=$fU;Dialstatus=$avp(s:dialstatus);User-Rate=$avp(i:2016);User-Duration=$avp(i:2016);User-Clid=$fU;User_ClidName=$avp(s:callerid);User_Dst_Name=$avp(s:2014);User_Dst_Code=$avp(i:2015);User_Account_Id=$fU;Lcr_Rate=$avp(i:2030);Lcr_Duration=$avp(i:2016);Lcrtrunk_Id=$avp(i:2027);Lcr_Dst_Name=$avp(s:2028);Lcr_Dst_Code=$avp(i:2029);Real_Duration=$avp(i:2016)") *dictionary.openser* ATTRIBUTE Dialstatus 1073 string ATTRIBUTE User-Rate 1074 string ATTRIBUTE User-Duration 1075 string ATTRIBUTE User-Channel 1076 string ATTRIBUTE User-Clid 1077 string ATTRIBUTE User_ClidName 1080 string ATTRIBUTE User_Dst_Name 1081 string ATTRIBUTE User_Dst_Code 1082 string ATTRIBUTE User_Account_Id 1083 string ATTRIBUTE Lcr_Rate 1084 string ATTRIBUTE Lcr_Duration 1085 string ATTRIBUTE Lcrtrunk_Id 1086 string ATTRIBUTE Lcr_Dst_Name 1087 string ATTRIBUTE Lcr_Dst_Code 1088 string ATTRIBUTE DialedTime 1089 string ATTRIBUTE Real_Duration 1090 string ATTRIBUTE Startleg 1091 string ATTRIBUTE Stopleg 1092 string *dictionary.radius* ATTRIBUTE Dialstatus 1073 string ATTRIBUTE User-Rate 1074 string ATTRIBUTE User-Duration 1075 string ATTRIBUTE User-Channel 1076 string ATTRIBUTE User-Clid 1077 string ATTRIBUTE User_ClidName 1080 string ATTRIBUTE User_Dst_Name 1081 string ATTRIBUTE User_Dst_Code 1082 string ATTRIBUTE User_Account_Id 1083 string ATTRIBUTE Lcr_Rate 1084 string ATTRIBUTE Lcr_Duration 1085 string ATTRIBUTE Lcrtrunk_Id 1086 string ATTRIBUTE Lcr_Dst_Name 1087 string ATTRIBUTE Lcr_Dst_Code 1088 string ATTRIBUTE DialedTime 1089 string ATTRIBUTE Real_Duration 1090 string ATTRIBUTE Startleg 1091 string ATTRIBUTE Stopleg 1092 string

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[OpenSER-Users] OpenSER Advanced Programming Course, Bucharest, Voice System, August 11-14, 2008
by Bogdan-Andrei Iancu 06 Aug '08

06 Aug '08

Hi everybody, Following the tradition started last year, the second edition of the "OpenSER Advanced Programming Course" will be held in Bucharest, Romania, starting with 11th of August 2008. The primary goal is the same: to attract more developers to OpenSER and build a higher development squad. There are three major upgrades from last edition: - free participation - there will be no fees for attending to this course; you just need to register. - there will be 4 days course instead of 3 days - the course will contain also a practical work where the students will have the chance to consolidate the knowledge they have accumulated during the theoretical presentations. The course will have a 60% - 40% distribution between theory and practice. The course is dedicated to people willing to learn how to develop OpenSER extensions. The course will follow the idea of writing a new openser module from scratch, with all the dependencies it has from the core: dynamic memory usage, MI interface, registering timers, locking, exporting pseudo-variables, etc The course is sponsored courtesy of : - Packt Publishing (donating 5% of the revenue from the Flavio E. Goncalves's "Building Telephony Systems with OpenSER" book.) - Voice System (logistics and hosting) Teachers from the OpenSER development team will be: - Bogdan-Andrei Iancu - Anca Vamanu As, said, the registration is free, but the class is limited as size up to 17 - 20 people. The seats will be granted on the "first in, first served" policy :). For registration, you need to mail me (bogdan(a)voice-system.ro) or to openser(a)voice-system.ro . You will receive a confirmation response to guarantee your registration. Best regards, Bogdan

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[Serusers] Textops problem
by Frank Hase 06 Aug '08

06 Aug '08

Hello, I´m using the SER for my phones at home. Now I want to phone as an anonymous. So I add a line in the ser.cfg: subst('/^From:(.*)sip[^@]*@[a- zA-Z0-9.]+(.*)$From:anonymous@anonymous.invalid;/ig'); But this doesn´t work. I just want to have in the Message Header the line From: anonymous(a)ananymous.invalid; So that I can phone anonym. Thank you for your help. Greeting Frank Haase

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Re: [Serusers] Serusers Digest, Vol 64, Issue 4
by Akintayo Olusegun 06 Aug '08

06 Aug '08

mail it again On Wed, Aug 6, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. 2.0 survey (Jiri Kuthan) > 2. ser 2.1 - compilation error - conflicting types > (Tomasz Zieleniewski) > 3. ser 2.1 - compilation error - conflicting types > (Tomasz Zieleniewski) > 4. SER/FIFO not accessible (cindy.cao) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 05 Aug 2008 19:20:33 +0200 > From: Jiri Kuthan <jiri(a)iptel.org> > Subject: [Serusers] 2.0 survey > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: <48988BE1.9030503(a)iptel.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I'm just wondering if folks using the 2.0 have found so far any issue > which has not been tracked -- I've been off-line for vacation, IETF and > a couple of other trips and began to be uncertain if I might have missed > something. Any concerns/issues/comments regarding 2.0 are most welcome, > publicly or privately. > > Thanks a lot > > -jiri > > > ------------------------------ > > Message: 2 > Date: Wed, 6 Aug 2008 01:11:34 +0200 > From: "Tomasz Zieleniewski" <tzieleniewski(a)gmail.com> > Subject: [Serusers] ser 2.1 - compilation error - conflicting types > To: serusers(a)lists.iptel.org, serusers(a)iptel.org > Message-ID: > <5fd52d7a0808051611g2017ff27uef0685ef960450d(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-2" > > Hi, > > When I try to compile ser from sources I get the following conflicting > types > errors: > gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops > -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' > -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' > -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.1"' -D__CPU_x86_64 -D__OS_linux > -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' > -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM > -DSHM_MMAP > -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE > -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST > -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK > -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM > -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD > -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM > -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT > -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lock_ops.c -o > lock_ops.o > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:46: error: conflicting types for 'loff_t' > /usr/include/linux/types.h:30: error: previous declaration of 'loff_t' was > here > /usr/include/sys/types.h:62: error: conflicting types for 'dev_t' > /usr/include/linux/types.h:13: error: previous declaration of 'dev_t' was > here > In file included from /usr/include/sys/types.h:133, > from ut.h:53, > from lock_ops.c:28: > /usr/include/time.h:105: error: conflicting types for 'timer_t' > /usr/include/linux/types.h:22: error: previous declaration of 'timer_t' was > here > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:198: error: conflicting types for 'int64_t' > /usr/include/linux/types.h:98: error: previous declaration of 'int64_t' was > here > /usr/include/sys/types.h:204: error: conflicting types for 'u_int64_t' > /usr/include/linux/types.h:97: error: previous declaration of 'u_int64_t' > was here > In file included from /usr/include/sys/types.h:220, > from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/select.h:78: error: conflicting types for 'fd_set' > /usr/include/linux/types.h:12: error: previous declaration of 'fd_set' was > here > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:235: error: conflicting types for 'blkcnt_t' > /usr/include/linux/types.h:124: error: previous declaration of 'blkcnt_t' > was here > make: *** [lock_ops.o] B??d 1 > > Is there any fix for that? > > Regards > Tomasz >

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[Kamailio-Users] Question about registrar lookup() and branching.
by Alex Balashov 06 Aug '08

06 Aug '08

Greetings, I've got a small problem with how registration lookups work. I have several concurrent registrations from one PBX that have multiple contact URIs that they declare that aren't the same; they contain the DIDs to route a call to. So, for instance, in Asterisk: register => user:pass@registrar/DID1 register => user:pass@registrar/DID2 Of course, lookup() doesn't care what those contacts actually are. If it sees multiple contacts for one username, t_relay() will simply create branches for all of them and and ring them all. I can't actually have that happen. I'm looking for suggestions on how to handle this most elegantly. The two options I've got in mind are: 1. Stop using lookup() and do my own URI rewrites and branch forking (if the contact URIs are identical) with attention to the contact. 2. Use BRANCH-ROUTEs to arrest the call leg if it is going to an RURI user part that does not match the user part of the contact URI, and... drop() the branches? I also need to build in some other logic, such that for example if a user has only one contact registered and it does not match the DID, or none of the contacts match the DID, proceed as normal anyway. Is it possible there is a better way? Recommendations welcome. Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Authentication problems.
by Buzukelis 06 Aug '08

06 Aug '08

*Hi all, ** I'm trying to register few ata's to openser.* *i puted two registrations to subscribers table.* *but registaration allwais unauthorized.* ** ** *ngrep output: * *interface: eth0 (192.168.27.0/255.255.255.0) filter: (ip) and ( port 5060 )* *U 192.168.27.117:10186 -> 192.168.27.201:5060 REGISTER sip:192.168.27.201 SIP/2.0. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK645148049;rport. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2000 REGISTER. Contact: *. Max-Forwards: 70. User-Agent: Grandstream HT-502 V1.0B 1.0.1.8. Expires: 0. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Length: 0. .* * U 192.168.27.201:5060 -> 192.168.27.117:10186 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK645148049;rport=10186. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b963. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2000 REGISTER. WWW-Authenticate: Digest realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.27.117:10186 -> 192.168.27.201:5060 REGISTER sip:192.168.27.201 SIP/2.0. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK142179605;rport. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2001 REGISTER. Contact: *. Authorization: Digest username="1000", realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7", uri="sip:19$ Max-Forwards: 70. User-Agent: Grandstream HT-502 V1.0B 1.0.1.8. Expires: 0. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Length: 0. . U 192.168.27.201:5060 -> 192.168.27.117:10186 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK142179605;rport=10186. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.3c26. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2001 REGISTER. WWW-Authenticate: Digest realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. openser.cfg :# # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # ------------------ module loading ---------------------------------- #set module path mpath="//lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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) # ------------------------- 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. if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; }; consume_credentials(); 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; } can not understand why this configuration is not working to me??? any help will be apreciated. *

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[Kamailio-Users] Change the INVITE timer after append_branch()
by Jens Carl 06 Aug '08

06 Aug '08

Hello List, I have call forwarding on no answer and the possibility to set the time out for the call. I do this by setting the value of "fr_inv_timer" with "avp_fr_timer_avp". Then when the call is route to user's phone and he has call forwarding on no answer activate I overwrite the default INVITE timer with his value. If the call is answered with the error message "Request Timeout (408)" I replace the RURI with the new number, call append_branch() and relay the message to myself. Now the message goes again to my routing logic and will be relayed. I also set a new timeout for the INVITE message, but still the value of the customer is used. For example if the users has a ring time of 10 seconds the second call also rings only for 10 seconds. How can I set the ringtime after append_branch() to my default value again? Is this possible or should I change me routing logic to handle this kind of feature? When I have to change it what is the best way to do this? Thanks in advance Jens

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[Kamailio-Users] 1.4.0 release plans, project matters
by Henning Westerholt 06 Aug '08

06 Aug '08

Hi all, Bogdan-Andrei Iancu prepared this fork from Kamailio (formerly known as OpenSER) since July. This was done secretly, we were not aware of this. We've all done a great amount of work to resolve the existing trademark issues and create a better organisational backing for the project. Many contributors tried hard to improve the quality of the code base, a big number of bug and documentation fixes were done, doxygen code documentation was greatly extended. Because of the project renaming our release schedule needed some adaptions. We discussed this in the last week on the developer list and found a new release date for Kamailio [1]. Version 1.4.0 will be released on the thursday, the 7. august. Cheers, Henning Westerholt [1] http://lists.kamailio.org/pipermail/devel/2008-July/014926.html

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[Serusers] SER/FIFO not accessible
by cindy.cao 06 Aug '08

06 Aug '08

2008-08-06 cindy.cao >Hello all: > >I am trying to add a user to mysql database with the command: > serctl add 1000 1000 1000(a)ser.com > but I get an error: SER/FIFO not accessible! >and i consulted some documents which said it should be the permission >problem. so i insert a line "fif0_mode=0777" into my "ser.cfg" and try the >command again, but now my system halt! > > Any pointers to where to start debuging this error? > Thanks so much > > Best Regards. >Cindy

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