- sr-users - kamailio.org

[Kamailio-Users] website updates
by Elena-Ramona Modroiu 07 Aug '08

07 Aug '08

Hi all, I have updated the website to reflect the renaming of the project. If you find broken links, please contact me. Best regards, Ramona

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[Kamailio-Users] includes in the route script ?
by Francois Berganz 07 Aug '08

07 Aug '08

is it possible to divide the script route in others files and do includes? With that, I want to have one file for route[1] , one for route[2] .

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Re: [Kamailio-Users] Users Digest, Vol 39, Issue 27
by luzango mfupe 07 Aug '08

07 Aug '08

Hi guys ..tired of this conflict..dont we supposed to have a mojor release today?? On 8/7/08, users-request(a)lists.kamailio.org <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: Question about registrar lookup() and branching. > (Gustavo Mistrinelli) > 2. Re: [Kamailio-Devel] [OpenSER-Users] OpenSER Advanced > Programming Course, Bucharest, Voice System, August 11-14, 2008 > (Andhika Gannesha Gemilang) > 3. Problem using additional attributes through radius_extra > parameter (Juha Heinanen) > 4. Re: [Kamailio-Devel] call for contribution: project logo > (Benny Prijono) > 5. Re: [Kamailio-Devel] call for contribution: project logo > (Torben Friese) > 6. Re: OpenSIPS project (Francois Menard) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 6 Aug 2008 09:59:43 -0300 > From: "Gustavo Mistrinelli" <gmistrinelli(a)gmail.com> > Subject: Re: [Kamailio-Users] Question about registrar lookup() and > branching. > To: "Alex Balashov" <abalashov(a)evaristesys.com> > Cc: users(a)lists.kamailio.org > Message-ID: > <c6717cc80808060559k24a7c774v96049c8fb5c7677a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Alex, > > I don't know your complete architecture, but let me ask you why Asterisk is > not a trusted gateway? > The you can use rewritehost or LCR logic to route DIDs to Asterisk > If you need to associate users with DIDs, you can use dbaliases. > > Cheers, > > Gustavo > On Tue, Aug 5, 2008 at 8:55 PM, Alex Balashov > <abalashov(a)evaristesys.com>wrote: > >> 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 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

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[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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[Kamailio-Users] call for contribution: project logo
by Daniel-Constantin Mierla 05 Aug '08

05 Aug '08

Hello, I already received the first proposal, courtesy of Darren Sessions. You can check at: http://www.kamailio.org/pub/kamailio-logos/ This message is an invitation for all of you to submit proposals for the new project logo. We are aware that the name is not making happy many people out there and we can work to solve it, maybe over the time will not look that bad or just send name suggestions. However, this should not stop you being creative and help the project, in a transitional or definitive state. Looking forward to your contributions. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] 2.0 survey
by Jiri Kuthan 05 Aug '08

05 Aug '08

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

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[Kamailio-Users] Openser version to run Mediaproxy 2.0
by Nuno Marques 05 Aug '08

05 Aug '08

Hello, I was reading the README file on the download page of Mediaproxy and found this statement: This version of MediaProxy (2.0 or higher) cannot be used in combination with any version of OpenSER older than 1.4 or any components of MediaProxy older than 2.0. You must completely upgrade any previous installation of OpenSER and MediaProxy to these new versions. What i want to ask you is if this is really true because Openser 1.4 (kamailio) isn't available yet... Can i use Mediaproxy 2.0 with Openser 1.3.2 or i need to wait for the Openser 1.4? Thanks in advance, Nuno Marques

2 1

[Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008
by Daniel-Constantin Mierla 05 Aug '08

05 Aug '08

Hello Everybody, as many expressed their intention for an openser/kamailio event in Bucharest, next week, I am going to organize a dinner in a place with nice Romanian beer, wine and food. Seeing the suggestions for the date, I am proposing August 11, 2008 (Monday) -- let me know if you have opinions. Exact hour and place will be sent as soon as I get an idea about number of participants. So far, expressed the intention to participate: - Henning Westerholt, developer and board member - Elena-Ramona Modroiu, developer and board member - Daniel-Constantin Mierla, developer and board member - Michiel ten Hagen - Sebastian Schuman Please write me if you want to attend, I need to know the number of attendants to make the reservation. The event is in the same fashion as the ones happening in Barcelona, Vienna, Berlin, ..., in the past -- everybody pays for its expenses. If you want to have some preamble (might not be same place, but...), you can check some photos from last similar event in Bucharest after the ROSDEV2008 conference: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/index1.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 05 Aug '08

05 Aug '08

David, After double checking, I do seem to be getting E.164 formatted numbers coming from the Cisco datasource, at least for the numbers we care about. Following is an example, you can see that I am able to successfully match a 1800 # as 1800, test (I just can't rate a price to it). id,StartTime,Username,CallerParty,Destination,DestinationId,DestinationName,Duration,Price,Gateway,KBIn,KBOut,Disconnect,Channel,Direction,CallType 1,2008-08-05 08:55:28,,5635833713,2415453,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/2:D:2,outgoing,Telephony 2,2008-08-05 08:55:28,,5635833713,2415453,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/0:D:1,incoming,Telephony 3,2008-08-05 08:55:11,,6082516000,19206750230,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/1:D:22,outgoing,Telephony 4,2008-08-05 08:55:11,,6082516000,19206750230,,,0,,,0,0,0,Normal release,,incoming,VoIP 5,2008-08-05 08:54:37,,6082888583,18004266918,1800,test,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/1:D:18,outgoing,Telephony On Fri, Aug 1, 2008 at 8:23 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > correct > > On Fri, Aug 1, 2008 at 8:11 PM, Brian Del Shasta <briandelshasta(a)gmail.com > > wrote: > >> Okay, I'm using a Cisco datasource, so I'm assuming that would replace >> OpenSER in the general schema, correct? >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[Kamailio-Users] Problem With SipTrace Module
by Ali Jawad 05 Aug '08

05 Aug '08

Hi I did enable the siptrace module to be able to see SIP packets in TLS mode. Once I did set it up, I could not logon to the server anymore. The log file does not show anything : Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=acc Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=nathelper Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=siptrace Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: connection type is xxxxxxx via TCP/IP Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: protocol version is 10 Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: server version is 5.0.48 Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=xlog Aug 5 10:28:21 [30793] DBG:xlog:child_init: init_child [-4] pid [30793] Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=acc Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=nathelper Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=siptrace And the server is listening correctly. # ----------------- SIP Trace Info modparam("siptrace", "db_url", "mysql://mysql://openser:xxxxxxxxx@xxxxxxxxxxxx/openser") modparam("siptrace", "trace_on", 1) Regards

2 2

[OpenSER-Users] XMPP/SIP-SIMPLE Problem
by Cesar Troncoso 05 Aug '08

05 Aug '08

Hi everyone, I have a problem with xmpp module. Messages going from xmpp to sip-simple work fine. The problem I have is in the opposite direction (from sip-sipmle to xmpp). If I set fork=no or children=1 in my config file then all messages are sent and received in both directions. This is what I see in the logs: xmpp: invalid content-type 0x300ff xmpp: unable to write to command pipe: Bad file descriptor Any Idea how to overcome this situation? Thanks! Cesar

2 5

[Kamailio-Users] Message authorization via XCAP
by Adrian Georgescu 05 Aug '08

05 Aug '08

I put this on OpenXCAP roadmap. Regards, Adrian Yann Derichard schrieb: > Hi, > > I have already put in place presence via XCAP and now I would like to > add instant messaging to my system so I would like to know if it is > possible to use OpenSER to do MESSAGE authorization via XCAP (using > OpenXCAP). Hmm - interesting question. It would be good to have a generic framework to check SIP requests against privacy rules stored in XCAP server. does anyone have more knowledge about this? regards klaus

1 0

Re: [Kamailio-Users] Debugging Openser v1.4 scripts
by Jim Burke 05 Aug '08

05 Aug '08

Hi Daniel, After a lot of trial and error I have worked it out! The following portion of script causes the error in both 1.3 and 1.4, however not in 1.2. Root cause is that I have not added "" in my case statements. I.e. case 404: should be case "404": The var reply_status is set from PV $rs in a on_reply or on_failure route using $var(reply_status) = $rs;. switch($(var(reply_status))) { case 404: #Not Found xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Terminating UA Not Found R4\n"); seturi("sip:rva404@domain"); $(avp(early_code)) = "404"; $(avp(early_reason)) = "Not Found!"; append_hf("P-App-name: early_announcement\r\n"); append_hf("P-App-Param: Final-Reply-Code=$(avp(early_code));Final-Reply-Reason=$(avp(early_reason)); "); break; SNIP default: sl_send_reply("$(var(reply_status))","$(var(reply_reason)))"); xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Default Route 4\n"); exit; break; } Let me know if you consider this a bug and I will open a ticket. Regards, Jim -----Original Message----- From: Jim Burke [mailto:jim@evolutiontel.net] Sent: Saturday, 2 August 2008 11:28 PM To: miconda(a)gmail.com Cc: users(a)lists.kamailio.org Subject: RE: [Kamailio-Users] Debugging Openser v1.4 scripts Hi Daniel, Have been busy, will send yuo the config shortly. For my information how do you plan to find the corrupt line. Is there something in GDB that can help us set a tracepoint and identify it? Regards, Jim -----Original Message----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Wednesday, 30 July 2008 12:03 AM To: Jim Burke Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] Debugging Openser v1.4 scripts Hello, On 07/29/08 15:43, Jim Burke wrote: > Hi All, > > We are working to get our script working on Version 1.4 and are having > problems debugging the script. Previously if I saw the following error in > the logs it would highlight the line and colomb numbers were the error > existed. Can someone help out? > > Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, > line 9459656, column 1-1: syntax error > Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, > line 9459656, column 1-1: bad command!) > Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) > it should still show the line and column where error occurred, but here is something wrong with the line. Can you obfuscate the private details in the config (passwords, ip addresses) and send it to me? Cheers, Daniel > Thanks, > Jim > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] Message authorization via XCAP
by Yann Derichard 05 Aug '08

05 Aug '08

Hi, I have already put in place presence via XCAP and now I would like to add instant messaging to my system so I would like to know if it is possible to use OpenSER to do MESSAGE authorization via XCAP (using OpenXCAP). Regards. -- Yann Derichard

2 1

[Kamailio-Users] Cluecon
by Bogdan-Andrei Iancu 05 Aug '08

05 Aug '08

Hi, For people interested in meeting and chatting around the kamilio - opensips subject, in the next 3 days I will be present at ClueCon, in Chicago. So feel free to drop me an email ;) Best regards, Bogdan

1 0

[Kamailio-Users] I don't know why
by Tseveendorj Ochirlantuu 05 Aug '08

05 Aug '08

Dear developers, I don't know what is the reason they will go to two ways. /---- Kamailio OpenSER \---- openSIPS I hope these two project going to powerful. We need OpenSER related projects. I like them. Best regards, Tseveendorj

1 0

[Kamailio-Users] TLS with openser
by Ali Jawad 04 Aug '08

04 Aug '08

hi all Got some issues with tls my pstn gw does not support tls ..but I am using it between my clients and openser. The traffic is being sent encrypted to the pstn gw. I am using sethostport to forward traffic. any ides ? Thx

3 4

[Serusers] cannot create /tmp/ ser_fifo
by 静于 04 Aug '08

04 Aug '08

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 a error: cannot create /tmp/ ser_fifo Any pointers to where to start debuging this error? Thanks. Best Regards. Ivy --------------------------------- 雅虎邮箱，您的终生邮箱！

2 1

[Kamailio-Users] INVITE branch handling for multiple contacts.
by Alex Balashov 03 Aug '08

03 Aug '08

Greetings, Is there a place somewhere that expounds upon or otherwise illustrates the inner workings of call forking for multiple concurrent contacts? I can only find a patchwork of hints by perusing the documentation for the 'tm' and 'registrar' modules, and they don't provide me entirely with the understanding that I am looking for. Failing that, I am seeking answers and clarifications to the following general questions. All of them concern handling of an INVITE in a situation where the registrar module allows multiple concurrent contacts, and multiple contacts are in fact registered, and the "append_branches" modparam to "registrar" is set to 1. 1. What is officially supposed to happen if multiple contacts to which the INVITE is relayed all answer the call instantaneously, whether via 183 early media or 200 OK -- but not OOB ringing events? What I get seems a little strange, although perhaps there is an explanation that I am missing. Both far-endpoints send a 200 OK w/SDP and I hear both media streams simultaneously when I call, interleaved by clicks. After a while, the second (most recent) contact's media stream drops off because the UAC decides that the call has timed out because it has not received ACK replies to its 200 OKs; at the same time, OpenSER appears to generate a CANCEL for that second call leg. The CANCEL is replied to with a 200 OK (not a 487 Request Terminated?) although this seems a little bizarre since the dialog's state is already established -- but since the 200 OKs are never replied to, I suppose it is not necessary to receive a BYE in order to terminate that request. Meanwhile, 200 OKs in response to INVITE keep coming from the contact that was slower to pick up and was CANCEL'd (why? this is Asterisk 1.4), but the ACKs from them keep being routed to the first contact (the one that remains), which must be understandably confused as to why they're there although processed as retransmissions. I have a packet capture I'd love to send someone for interpretation, but would rather do it privately off-list. 2. Is there a shared REPLY-ROUTE for replies on each branch? Or is there a way to fragment particular reply routes for each branch? The goal I am ultimately looking to achieve is surely commonplace enough; I would like multiple concurrent registrants to be able to be rung at one request URI. But at present, it seems some very strange things are happening when this occurs that confuse the far-end UACs. That's what governs my intuition that I just need to properly understand how branching is supposed to work. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 6

[Kamailio-Users] multidomain test for domains 192.168.0.3 and 192.168.0.32
by Zen Kato 03 Aug '08

03 Aug '08

I have installed openser-1.3.2-nontls on 192.168.0.3. The phones are 0103(a)192.168.0.3, 0203(a)192.168.0.3 and 0303(a)192.168.0.3. To test multidomain support, I added modparam("auth_db|auth_diameter|group|uri_db|usrloc","use_domain",1) and changed from "uri==myself" to "is_uri_host_local" in openser.cfg. I added two phones 9101 and 9102 with domain 192.168.0.32. These two phones are registered on mysql db as follows; # openserctl add 9101(a)192.168.0.32 password 9101@localhost # openserctl add 9102(a)192.168.0.32 password 9102@localhost I also set up 192.168.0.32 for eth0:0. eth0 is 192.168.0.3. Then, I check: [root@p4c8 multidomain]# openserctl domain show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd domain_dump 192.168.0.3 192.168.0.32 FIFO command was: :domain_dump:openser_receiver_6329 [root@p4c8 multidomain]# openserctl domain showdb database engine 'MYSQL' loaded Control engine 'FIFO' loaded +----+--------------+---------------------+---------+ | id | domain | last_modified | version | +----+--------------+---------------------+---------+ | 2 | 192.168.0.3 | 2008-04-09 16:13:12 | 0 | | 4 | 192.168.0.32 | 2008-01-01 00:00:00 | NULL | +----+--------------+---------------------+---------+ [root@p4c8 multidomain]# openserctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=3 max_slot=1 AOR:: 0203(a)192.168.0.3 Contact:: sip:0203@192.168.0.102 Q= Expires:: 295 Callid:: 91741a3a146e34bb(a)192.168.0.102 Cseq:: 101 User-agent:: Grandstream BT110 1.0.8.23 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 AOR:: 0303(a)192.168.0.3 Contact:: sip:0303@192.168.0.103 Q= Expires:: 294 Callid:: 4c0a9450bb744c33(a)192.168.0.103 Cseq:: 101 User-agent:: Grandstream BT100 1.0.6.8 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 AOR:: 0103(a)192.168.0.3 Contact:: sip:0103@192.168.0.101 Q= Expires:: 295 Callid:: 60507d20c32ac791(a)192.168.0.101 Cseq:: 101 User-agent:: Grandstream BT110 1.0.8.23 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 FIFO command was: :ul_dump:openser_receiver_6424 openserctl ul show does not show 9101(a)192.168.0.32 and 9102(a)192.168.0.32 !? [root@p4c8 multidomain]# openserctl ul show 9101(a)192.168.0.32 database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_show_contact location 9101(a)192.168.0.32 404 AOR not found FIFO command was: :ul_show_contact:openser_receiver_6394 location 9101(a)192.168.0.32 What else shoud I set to test multidomain in this situation? Regards, Zen

1 1

[Kamailio-Users] error adding users using openserctl
by muhammad akl 03 Aug '08

03 Aug '08

hey all , i have successfuly built openser-1.2.2-tls from source and till now all things seems to be fine faced a problem with openser_mysql.sh and i have solved it but when i tried to add users to the DB using openserctl i got this message : database engine 'MYSQL' loaded Control engine 'FIFO' loaded /sbin/openserctl: no: not found /sbin/openserctl: no: not found usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] is_user: user counter= Error: user 'zzzz' already exist however i havn't added any users before so any help are more welcomed BR muhammad

1 1

[Serusers] LCR question
by Deepak 03 Aug '08

03 Aug '08

Hi can someone please explain me how LCR works. Specifically, scenario: prefix 1514 handled by gw_1 and gw_2 all other prefixes handled by gw_3 1) Call comes w/ RURI = 1514111111. 2) load_gws is called. In the database, prefix 1514 is supposed to be handled by gw_1 and gw_2; 3) next_gw() is called. RURI is now 1541111111@gw_1 4) call routed to gw_1; gw_1 rejects call 5) we go to failure route and call load_gws() again -- [QUESTION: what happens here?? how does SER know that we need to replace URI w/ 151411111@gw_2 ] 6) next_gws() called. I assume call will go to 1541111111@gw_2 7) gw_2 rejects call. We again go to failure_route. Here again we do a load_gws(). [QUESTION: How does SER know that both gw_1 and gw_2 options have been utilized?? Does load_gws return an error here or does it load gw_3? ] My problem is that in my case, gw_3 is being used after gw_1 and gw_2 have been used. As per my understanding this should NOT happen since prefix 151411111 is not handled by gw_3. Any pointers appreciated.. Running ser 0.9.6

1 0

[Kamailio-Users] Debugging Openser v1.4 scripts
by Jim Burke 02 Aug '08

02 Aug '08

Hi All, We are working to get our script working on Version 1.4 and are having problems debugging the script. Previously if I saw the following error in the logs it would highlight the line and colomb numbers were the error existed. Can someone help out? Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: syntax error Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: bad command!) Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) Thanks, Jim

2 2

[Kamailio-Users] T.38 not relayed by RTPproxy
by Jens Carl 02 Aug '08

02 Aug '08

Hello List, I have problem with T.38 and the RTPproxy by one of mine scenarios. The first one works flawless and looks like that: Linksys ATA -> NAT -> OpenSer/RTPproxy -> T.38-GW (Asterisk) The second one on the other hand don't work how it should and looks like this one: T.38-GW -> SIP-Proxy -> Openser/RTPproxy -> NAT -> Linksys ATA. The normal RTP stream is opened and they switch over to T.38, but the RTPproxy isn't sending out the received T.38 packets from the ATA. If the OpenSER is getting a re-INVITE I call the function force_rtp_proxy(). I already tried the function with 'l' parameter but then none of the scenarios are working. Why is the RTPproxy not relaying like in the 1st scenario? Which flag should I use to get the things working? Why is RTPproxy using the same ports like it used for the normal RTP traffic? I thought when I call force_rtp_proxy() a 2nd time I create a new session with different ports? Can I force the RTPproxy to use different ports? If someone needs a PCAP-trace I can provide this. Many thanks Jens

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Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 02 Aug '08

02 Aug '08

I see. So in my case, since I'm using freeradius, I would need to instruct freeradius to send out the call in E.164 format? I'll look into that... On Fri, Aug 1, 2008 at 3:45 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > I mean instruct OPENSER to send out the call like that, and in cdrtool set > the areacode 1305 and description usa-miami. I'll get you the tables, but > I'll need a couple of days as I'm on the road right now. > > d > > > On Fri, Aug 1, 2008 at 2:44 AM, Brian Del Shasta <briandelshasta(a)gmail.com > > wrote: > >> By "sending out the call in E.164 format" do you mean instructing the >> gateway (or something other than CDRTool) to use that format? And are you >> saying to set the destination id to '1305 USA-MIAMI' or to just '1305' and >> the description to USA-MIAMI? I don't suppose by any chance you have access >> to still and can include an example line from your Destinations, Customers, >> Profiles, and Rates tables? I appreciate your help! >> >> >> On Thu, Jul 31, 2008 at 10:04 PM, David Villasmil < >> david.villasmil.work(a)gmail.com> wrote: >> >>> The only way I got it to work was by sending out the call in E.164 >>> format, i.e. +1035 and setting that same areacode in cdrtool like: 1305 >>> USA-MIAMI... remember pricing is done like 10000 for 1 dolar/euro >>> >>> david >>> >>> On Thu, Jul 31, 2008 at 10:55 PM, Brian Del Shasta < >>> briandelshasta(a)gmail.com> wrote: >>> >>>> >>>> I'm sorry, I don't understand. Do you mean within CDRTool or like on >>>> our actual gateway which feeds info to the freeradius db? >>>> >>>> On Thu, Jul 31, 2008 at 4:41 PM, David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> You MUST set the dialed numbers as E.164 ( I think that's the number ), >>>>> like +1305 for Miami, US. or else you'll never get it working... >>>>> >>>>> d >>>>> >>>>> >>>>> On Thu, Jul 31, 2008 at 11:47 AM, Brian Del Shasta < >>>>> briandelshasta(a)gmail.com> wrote: >>>>> >>>>>> Ok, I changed the time zone to America/Chicago. If anyone else has >>>>>> any luck setting up ratings within CDRTool, I would love to speak with you, >>>>>> thanks... >>>>>> >>>>>> >>>>>> On Thu, Jul 31, 2008 at 10:18 AM, Adrian Georgescu < >>>>>> ag(a)ag-projects.com> wrote: >>>>>> >>>>>>> Central Time is not s valid timezone >>>>>>> Further than this your rates look ok at first glance, if it does not >>>>>>> work maybe other can help with their experiences. >>>>>>> Adrian >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users(a)lists.kamailio.org >>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)lists.kamailio.org >>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)lists.kamailio.org >>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

2 2

[Kamailio-Users] enum module in openser
by Ali Soltani 01 Aug '08

01 Aug '08

Hi, I am using Enum Module in opencer. Here is the piece of the code I am using in openser.cfg: route { ############################################################## if (method=="INVITE") { xlog ("This is the ReqUri as it arrives to openser: $ru"); # before adding + subst_uri ('/^sip:([0-9]+)/sip:+\1/'); xlog ("This is the ReqUri as it arrives to openser: $ru"); # after adding + enum_query("labenum.e164.arpa.com","10.0.0.1"); xlog("This is the return code from the enum query: $rc"); # returns -1 xlog("This is your new ReqUri: $ru"); # no URI change ..... } } Even though the response from Enum query is being sent to openser, enum_query returns -1. enum_query also does not change the current URI. here is the format of enum entry in the dns server: # dig 0.0.0.1.4.8.4.3.1.8.labenum.e164.arpa.com @10.0.0.1 +short 100 10 "U" "E2U+sip" "!^.*$!sip:+18134841000@bts.voice.net!" . Any suggestion? Thanks,

2 1

[Kamailio-Users] About TLS
by Ali Jawad 01 Aug '08

01 Aug '08

Hi All I have Openser running on both tls and udp. On upd it works fine. However when I do enable TLS it connects and I can make call however when I close the call from the softphone the cell phone keeps ringing. If I do close the call from the cell phone the softphone redials the number and calls the cellphone again. I know you guys might want to see a trace for the call however on the server side I did execute : ngrep port 443 and host 87.236.144.9 .......f.\..o..g.g`....W.^%$jp3.TI...zi...f7..8........j.....KV2^>.....M JG.D........S..M..bF...2..qn..b..7g.).J:....!...&.;P?&<..X....v.z.:._.[. v...JMt.4....A%Q...K..W...aD?R ..=."=v.R...m.5G...(V..k6..si.....'.. 9<X.+.....y..P.#..[..GI(<..$<.......4."#.L[....{..Z.......$h..t3zy<.=... . pJ.../......^y.lO.i &\.Q..aA......j('.j.v...{6..m.S.........J..;.6....!......?..W.o...{*.... |..c7...OaTR...O.]R....=t4...m|d..]....7.6w&\.S..62..N.S.T.6m.2.]..x|9]. 2..]r...C..'.zW.,.....z#..l.S.9.I.JD-1.j .q..%.......t..."C..L3...Y!%3$.....)..y.. As you can see the output is encrypted. Shouldn't the output be clear text ? Thanks

3 2

Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 01 Aug '08

01 Aug '08

By "sending out the call in E.164 format" do you mean instructing the gateway (or something other than CDRTool) to use that format? And are you saying to set the destination id to '1305 USA-MIAMI' or to just '1305' and the description to USA-MIAMI? I don't suppose by any chance you have access to still and can include an example line from your Destinations, Customers, Profiles, and Rates tables? I appreciate your help! On Thu, Jul 31, 2008 at 10:04 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > The only way I got it to work was by sending out the call in E.164 format, > i.e. +1035 and setting that same areacode in cdrtool like: 1305 USA-MIAMI... > remember pricing is done like 10000 for 1 dolar/euro > > david > > On Thu, Jul 31, 2008 at 10:55 PM, Brian Del Shasta < > briandelshasta(a)gmail.com> wrote: > >> >> I'm sorry, I don't understand. Do you mean within CDRTool or like on our >> actual gateway which feeds info to the freeradius db? >> >> On Thu, Jul 31, 2008 at 4:41 PM, David Villasmil < >> david.villasmil.work(a)gmail.com> wrote: >> >>> You MUST set the dialed numbers as E.164 ( I think that's the number ), >>> like +1305 for Miami, US. or else you'll never get it working... >>> >>> d >>> >>> >>> On Thu, Jul 31, 2008 at 11:47 AM, Brian Del Shasta < >>> briandelshasta(a)gmail.com> wrote: >>> >>>> Ok, I changed the time zone to America/Chicago. If anyone else has any >>>> luck setting up ratings within CDRTool, I would love to speak with you, >>>> thanks... >>>> >>>> >>>> On Thu, Jul 31, 2008 at 10:18 AM, Adrian Georgescu <ag(a)ag-projects.com>wrote: >>>> >>>>> Central Time is not s valid timezone >>>>> Further than this your rates look ok at first glance, if it does not >>>>> work maybe other can help with their experiences. >>>>> Adrian >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)lists.kamailio.org >>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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[OpenSER-Users] Need help on Openser + freeradius
by ha do 01 Aug '08

01 Aug '08

Hi i am trying to config Openser to talk to Freeradius i install openser on first PC : 192.168.1.4 i install freeradius on second PC : 192.168.1.5 on the Openser, i change: edit /etc/radiusclient-ng/radiusclients.conf : authserver & accserver 192.168.1.5 edit etc/radiusclient-ng/servers : 192.168.1.5 openser on the FreeRadius, i change: edit clients.conf : listen { 192.168.1.4 ....} but when i make call Openser doesnt talk to FreeRadius does any ideal on it my openser config: modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("auth_db|permissions|uri_db|usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("nathelper","rtpproxy_disable", 1) modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") oute[3] { ## Requests handler if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); # Radius Extra $avp(s:sip_proxy_ip)="192.168.1.5"; }; Thanks

3 2

[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 01 Aug '08

01 Aug '08

Central Time is not s valid timezone Further than this your rates look ok at first glance, if it does not work maybe other can help with their experiences. Adrian

3 3

[Kamailio-Users] Patch for mi_xmlrpc module Makefile
by Sergio Gutierrez 31 Jul '08

31 Jul '08

Hello everybody. A patch has been published on tracker, which fixes a problem related to not define a macro which makes fail the module compilation in Solaris. It has been registered with Request ID: 2034302* * Hope it be useful. Best regards. Sergio Gutiérrez.* *

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[Kamailio-Users] Kamailio Logo
by Darren Sessions 31 Jul '08

31 Jul '08

Off technical topic, I know, but has the Kamailio logo already been designed? - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com http://www.linkedin.com/in/dsessions _____________________________

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[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

You need to add at list one customer entry, associated profiles <----

2 1

[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

I am afraid that rating all calls the same regardless of destination will not work unless you define some destination to match all your traffic. The rating engine must find a destination prefix from which it derives the price. Read the RATING.txt to understand the logic behind it. Adrian >> I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination...

2 1

Re: [Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

Yes You need to add at list one customer entry, associated profiles and finally the rates linked with the profile which contain the actual prices for each destination. You have example rating files that you can modify. Adrian On Jul 31, 2008, at 3:28 PM, Brian Del Shasta wrote: > That's not a problem. I am currently able to successfully match > destinations by looking for anything that starts with a '1'. I have > a destination setup called "Billable" which applies to any > destination beginning with 1. I also have destinations setup for > 1800, 1877, etc to filter out the 800 numbers. > > What I cannot figure out is how to link a rate to a destination. Do > I have to setup a profile and customer as well? Or should I be able > to price calls using only destination and rates? > > On Thu, Jul 31, 2008 at 8:42 AM, Adrian Georgescu <ag@ag- > projects.com> wrote: > I am afraid that rating all calls the same regardless of destination > will not work unless you define some destination to match all your > traffic. > The rating engine must find a destination prefix from which it derives > the price. Read the RATING.txt to understand the logic behind it. > Adrian > >> > I'm attempting to understand how call rating works within CDRTool. > I am > able to link destinations to the calls, but unable to generate > prices. I'm > using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What > I'm > trying to setup seems simple: I have one set rate of $.08/min for any > call > beginning with a 1 (I'll exclude 800 numbers later). This rate should > be > applied globally with no differentiation between caller party or > destination... > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

Hi Neill, The outgoing audio works fine. The audio that is not passed is from PSTN -> Phone. The Snom does have symmetrical RTP set. I did have this working about a week ago then it stopped working! How does asterisk deal with NAT, as if I connect the phone directly to Asterisk everything works perfectly! Both Asterisk & Kamailio are on publically accessible IP addresses. Thank you for your help so far. Ross From: neill.wilkinson(a)btinternet.comTo: users(a)lists.kamailio.orgDate: Thu, 31 Jul 2008 13:32:05 +0100Subject: Re: [Kamailio-Users] NAT Problems Which direction is the one way audio in – from the Phone to the PSTN or Vice-Versa – I assume audio outbound is OK and it’s the audio returning to the phone from the PSTN you’re missing? Do you have symmetrical RTP set in the Line settings (under NAT) on the SNOM? Is Kamailio inside or outside the NAT? Neill...;o) Neill WilkinsonPrincipal Consultant Aeonvista Ltd - opening up new ideas From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ross BeerSent: 31 July 2008 12:50To: users(a)lists.kamailio.orgSubject: [Kamailio-Users] NAT Problems I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= 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;};# NAT detectionforce_rport();if(nat_uac_test("3")){fix_nated_contact();}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 (uri==myself) {if (method=="REGISTER") {if (!www_authorize("", "subscriber")) {www_challenge("", "0");exit;};if (isflagset(5)) {# set branch flag -- when someone will call this user# the INVITE will have branch flag 6 set after lookup("location")setbflag(6); };fix_nated_contact();fix_nated_register();consume_credentials();save("location");exit;};if (!lookup("location")) {sl_send_reply("404", "Not Found");exit;};append_hf("P-hint: usrloc applied\r\n"); };if(method=="MESSAGE"){if (!lookup("location")){sl_send_reply("404", "User Offline");exit;}route(4);}if (method=="REGISTER"){save("location");} route(1);} route[1] {if (subst_uri('/(sip:.*);nat=yes/\1/')){setbflag(6);};if (isflagset(5)||isbflagset(6)) {route(3);}t_on_reply("1");if (!t_relay()) {sl_reply_error();};exit;} route[3] {if (is_method("BYE")) {unforce_rtp_proxy();} else if (is_method("INVITE")){fix_nated_sdp("3"); };if (isflagset(5)){search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');}t_on_reply("1");}route[4]{if (!t_relay()) {sl_reply_error();};exit;} failure_route[2] {if (isbflagset(6) || isflagset(5)) {unforce_rtp_proxy();}}onreply_route[1] {fix_nated_contact();if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) {fix_nated_sdp("2");force_rport(); }search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');} ==================== SIP PACKETS ===========================INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asteriskMax-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxxs=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY ////////////////////SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID ***CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 Find out how to make Messenger your very own TV! Try it Now! _________________________________________________________________ Invite your Facebook friends to chat on Messenger http://clk.atdmt.com/UKM/go/101719649/direct/01/

2 1

[Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= 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; }; # NAT detection force_rport(); if(nat_uac_test("3")) { fix_nated_contact(); } 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 (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # the INVITE will have branch flag 6 set after lookup("location") setbflag(6); }; fix_nated_contact(); fix_nated_register(); consume_credentials(); save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; if(method=="MESSAGE") { if (!lookup("location")) { sl_send_reply("404", "User Offline"); exit; } route(4); } if (method=="REGISTER") { save("location"); } route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { fix_nated_sdp("3"); }; if (isflagset(5)) { search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("1"); } route[4] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[2] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { fix_nated_contact(); if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) { fix_nated_sdp("2"); force_rport(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } ==================== SIP PACKETS =========================== INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asterisk Max-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxx s=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY //////////////////// SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID *** CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508 v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP *** t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 _________________________________________________________________ Find the best and worst places on the planet http://clk.atdmt.com/UKM/go/101719807/direct/01/

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[Kamailio-Users] Kamailio and Snom phones
by Jason Penton 31 Jul '08

31 Jul '08

Hi * not sure if someone has encountered this before but I have noticed a problem with Snom phones registering with Kamailio. It seems that the Snom phones register with a contact port of 2049. However, when Kamailio does the usrloc lookup and the corresponding t_relay, a packet dump shows me that Kamailio is trying to send the invite to port 5060. Can anyone shed some light? Kind regards Jason

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[Kamailio-Users] Error in database module
by Schumann Sebastian 31 Jul '08

31 Jul '08

Dear all I use OpenSER with the perl extension to publish presence states for non-SIMPLE/PUBLISH capable devices using an outbound proxy. My idea is to send all SIP messages to an outbound-proxy, that forwards them to the respective registrar/SIP proxy. The outbound proxy uses SIP messages (INVITE, BYE, CANCEL...) as triggers to call a perl script that publishes the presence state to a presence agent using PUA_MI. Currently, registrar, presence agent and the publishing functionality are hosted on one machine. But I think that should not have any impact on the whole idea... All what I wanted to do works fine so far, except that after some time, I get MySQL errors. The Perl script does not use the database, only from OpenSER script, actions are called. I posted below some of the error messages I receive and after, the part of my config performing database queries. Does anyone see a obvious mistake in there? I tried OpenSER lastest branch and current trunk, both produce the error after some time. Thanks for your help! Sebastian Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:pua:db_update: while querying db table pua Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:xcap_client:query_xcap_update: in sql query Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_delete: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:update_db_subs: deleting expired information from database Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:msg_watchers_clean: querying database for expired messages Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:msg_presentity_clean: querying database for expired messages Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:pua:db_update: while querying db table pua Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:xcap_client:query_xcap_update: in sql query Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_delete: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:update_db_subs: deleting expired information from database Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:msg_watchers_clean: querying database for expired messages Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:msg_presentity_clean: querying database for expired messages 203 if (uri==myself) { 204 if (method=="REGISTER") { 205 # Uncomment this if you want to use digest authentication 206 if (!www_authorize("IP", "subscriber")) { 207 www_challenge("IP", "0"); 208 exit; 209 }; 210 211 # query if dialog exists for registering URI 212 avp_delete("*"); 213 if (avp_db_query("SELECT from_uri, 214 to_uri 215 FROM dialog 216 WHERE from_uri='$tu' or to_uri='$tu' or from_uri='$tu:5060' or to_uri='$tu:5060'")) { 217 #xlog("L_INFO", ">>> Dialog active: caller=$avp(i:1) callee=$avp(i:2)\n"); 218 } else { 219 #xlog("L_INFO", ">>> No Dialog active: avp_db_query() returned no records\n"); 220 }; 221 xlog("L_INFO", ">>>>>>>>> REGISTER received ($tu)\n"); 222 if(!is_avp_set("$avp(i:1)")) { # no call active for registering to_uri user 223 xlog("L_INFO", ">>>>>>>>> PUBLISHED\n"); 224 perl_exec("publish"); 225 } else { 226 xlog("L_INFO", ">>>>>>>>> NOT PUBLISHED, call active\n"); 227 } 228 save("location"); 229 exit; 230 } 231 232 if(is_method("SUBSCRIBE|PUBLISH")) { 233 if(is_method("PUBLISH")) { 234 if($si == "IP") { 235 if (!t_newtran()) { 236 sl_reply_error(); 237 exit; 238 }; 239 handle_publish(); 240 t_release(); 241 } else { 242 sl_send_reply("488","Not Acceptable Here"); 243 } 244 } else if(is_method("SUBSCRIBE")) { 245 if (!proxy_authorize("", "subscriber")) { 246 proxy_challenge("", "1"); 247 exit; 248 }; 249 handle_subscribe(); 250 t_release(); 251 }; 252 exit; 253 }; 252 exit; 253 }; 254 255 lookup("aliases"); 256 257 # native SIP destinations are handled using our USRLOC DB 258 if (!lookup("location")) { 259 sl_send_reply("404", "Not Found"); 260 exit; 261 } else { 262 append_hf("P-hint: usrloc applied\r\n"); 263 if(method=="INVITE") { 264 # Store Dialog in db 265 setflag(5); 266 # Session timer 267 setflag(6); 268 if (sstCheckMin("1")) { 269 xlog("L_ERR", "422 Session Timer Too Small reply sent.\n"); 270 exit; 271 } 272 xlog("L_INFO", ">>>>>>>>> INVITE received ($fu busy)\n"); 273 perl_exec("publish"); 274 } 275 }; 276 };

3 5

[Kamailio-Users] FYI: A1 InnovationDays
by Daniel-Constantin Mierla 31 Jul '08

31 Jul '08

Maybe some developer has a brilliant idea involving openser and looks for some support. Also, maybe you will find there ideas that will help you in the future, so you can comment and vote for it. Cheers, Daniel :: Final Call for the International Developer Challenge A1 InnovationDays :: Since almost 4 weeks, the international developer challenge A1 InnovationDays attracted more than 30 ideas from creative minds. The challenge is open to anyone who addresses the grand theme "Open Communication by Open Standards" and can come up with ideas for cool services built on IP-based communication (SIP & IMS). Ideas can be posted on the platform www.a1innovations.at/en/static/a1innovationdays until August 15, 2008. All entries are open for voting and discussing by all users. On August 15, the 10 highest ranked ideas will be short-listed and judged by a panel of recognized experts as well as representatives from the sponsors. End of August, the best teams behind the ideas will be invited to a prototyping event near Vienna in order to realize their ideas and demonstrate their potential. The winning team can choose among several awards, such as a week of powder snow skiing in Austria. After that, the sponsor (mobikom Austria) is keen to bring your idea to the market. All submissions must be based on Open Source Software. : More Information : For all details about this challenge, go to the FAQs: http://www.a1innovations.at/en/static/faq For registration, go directly to: http://www.a1innovations.at/registrierung : Important dates ahead : August 1, 2008: Two prizes for the best interim ideas will be awarded August 15, 2008: End of idea entries (CET 24:00) August 18, 2008: Announcement of the five selected teams for the prototyping event August 29 - September 1, 2008: Prototyping event at the monastery Und (near Vienna: http://www.und.at/) Other notes about the event: http://blog.tmcnet.com/blog/tom-keating/voip/a1-innovationdays-contest-look… -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Detecting Two Phones Behind Same NAT
by Ross Beer 31 Jul '08

31 Jul '08

Hi, Is there a way to detect two phones behind the same NAT device? Thanks in advance, Ross _________________________________________________________________ The John Lewis Clearance - save up to 50% with FREE delivery http://clk.atdmt.com/UKM/go/101719806/direct/01/

3 3

[OpenSER-Users] Firewall
by CSB 31 Jul '08

31 Jul '08

I have an error which is driving me crazy: Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:core:udp_send: sendto(sock,0x81aaed8,825,0,0xb61216f0,16): Operation not permitted(1) Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:tm:msg_send: udp_send failed This is firewall related because when I turn iptables off the problem goes away. But although all ACCEPT and DENY messages are logged by the firewall, there is no corresponding message logged. When a call is made between two UACs and the far end attempts to answer the call, the call is not answered successfully. The dialogue is shown below. If I turn the firewall off, start a call and then turn the firewall on the call continues successfully. However the next call is not successful. We use Mediaproxy and I suspect that is trying to do some communication that is being blocked by the firewall but that is only a suspicion. The dialogue for this call is shown at the bottom. When comparing these two dialogues with the firewall turned on: - when the callee answers, OpenSER receives a 200 OK but doesn't immediately pass that on to the caller - after a number of 200 messages from the callee, OpenSER sends the 200 to the caller but the Contact is the private IP address whereas when the firewall is turned off it is the public IP address I have also included the iptables config. Note that some lines are commented out due to the testing I'm doing and IP addresses have been changed. Any advice on the changes required would be appreciated. Thanks **************************** SIP dialogue with firewall on U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2222 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.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D;rport=5060;received=58.28.001.0 01..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35982 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.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 192.168.1.124:5065ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=566CA8D1-4C8E0458>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK4b4d0d4aF803AD55..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************************** SIP dialogue with Firewall off U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2224 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.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366;rport=5060;received=58.28.001.00 1..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 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.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=3D7BF99A-F3B3ACE5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.2..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************** iptables config $IPTABLES -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A OUTPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT # # Rule 0 (tun0,tun1,tun2) # echo "Rule 0 (tun0,tun1,tun2)" # # # $IPTABLES -N In_RULE_0 $IPTABLES -A INPUT -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A In_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A In_RULE_0 -j ACCEPT $IPTABLES -N Out_RULE_0 $IPTABLES -A OUTPUT -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A Out_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A Out_RULE_0 -j ACCEPT $IPTABLES -A Out_RULE_0 -j ACCEPT # # Rule 1 (lo) # echo "Rule 1 (lo)" # # # $IPTABLES -N In_RULE_1 $IPTABLES -A INPUT -i lo -p icmp -m icmp --icmp-type any -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j In_RULE_1 #$IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_1 $IPTABLES -A In_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A In_RULE_1 -j ACCEPT $IPTABLES -N Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p icmp -m icmp --icmp-type any -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j Out_RULE_1 #$IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_1 $IPTABLES -A Out_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A Out_RULE_1 -j ACCEPT # # Rule 2 (eth0) # echo "Rule 2 (eth0)" # # # $IPTABLES -N Out_RULE_2 $IPTABLES -A OUTPUT -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A FORWARD -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A Out_RULE_2 -j LOG --log-level warning --log-prefix "RULE 2 -- ACCEPT " $IPTABLES -A Out_RULE_2 -j ACCEPT # # Rule 3 (eth0) # echo "Rule 3 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E561900.0 $IPTABLES -A INPUT -i eth0 -d 147.202.001.001 -m state --state NEW -j Cid485B0E561900.0 $IPTABLES -N Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E561900.1 $IPTABLES -N In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 203.89.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 58.28.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 64.38.001.001 -j In_RULE_3 $IPTABLES -A In_RULE_3 -j LOG --log-level warning --log-prefix "RULE 3 -- ACCEPT " $IPTABLES -A In_RULE_3 -j ACCEPT # echo "Rule 4 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E6A1900.0 $IPTABLES -A OUTPUT -o eth0 -s 147.202.001.001 -m state --state NEW -j Cid485B0E6A1900.0 $IPTABLES -N Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E6A1900.1 $IPTABLES -N Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 203.89.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 58.28.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 64.38.001.001 -j Out_RULE_4 $IPTABLES -A Out_RULE_4 -j LOG --log-level warning --log-prefix "RULE 4 -- ACCEPT " $IPTABLES -A Out_RULE_4 -j ACCEPT # # Rule 5 (eth0) # echo "Rule 5 (eth0)" # # # $IPTABLES -N In_RULE_5 $IPTABLES -A INPUT -i eth0 -p tcp -m tcp -m multiport -d 147.202.001.001 --dports 5060,22,443,80,53,25060,8008 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 10000:20000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 35000:36000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport -d 147.202.001.001 --dports 1194,5065,5060,53,10000:20000,35000:36000 -m state --state NEW -j In_RULE_5 $IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_5 $IPTABLES -A In_RULE_5 -j LOG --log-level warning --log-prefix "RULE 5 -- ACCEPT " $IPTABLES -A In_RULE_5 -j ACCEPT # # Rule 6 (eth0) # echo "Rule 6 (eth0)" # # # $IPTABLES -N Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p tcp -m tcp -m multiport -s 147.202.001.001 --dports 22,53,80,443,5060,8008,25060,25 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 10000:20000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 35000:36000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport -s 147.202.001.001 --dports 53,1194,5060,5065,1813,123,10000:20000,35000:36000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A Out_RULE_6 -j LOG --log-level warning --log-prefix "RULE 6 -- ACCEPT " $IPTABLES -A Out_RULE_6 -j ACCEPT # # Rule 7 (global) # echo "Rule 7 (global)" # # # $IPTABLES -N RULE_7 $IPTABLES -A OUTPUT -j RULE_7 $IPTABLES -A INPUT -j RULE_7 $IPTABLES -A FORWARD -j RULE_7 $IPTABLES -A RULE_7 -j LOG --log-level warning --log-prefix "RULE 7 -- DENY " $IPTABLES -A RULE_7 -j DROP # #

2 1

[Kamailio-Users] wildcard in avp_db_load with db_schema
by Andreas Granig 31 Jul '08

31 Jul '08

Hi, For loading all AVPs from "normal" avp tables, I can use something like this: avp_db_load("$fu/domain", "*/domain_preferences"); But when I want to use a db schema to load AVPs belonging for example to a call_id for some sort of dialog tracking, it fails to start with the following error message: "ERROR:avpops:parse_avp_db: inconsistent usage of DB scheme without complet specification of AVP name" My config looks like that: modparam("avpops","db_scheme", "my_schema:table=some_table;uuid_col=call_id;value_col=value") ... avp_db_load("$ci", "*/$my_schema"); If I put something like "$avp(i:10)" instead of the "*", it works, but then I just get this specific AVP instead of all of them. Any idea if this is supposed to work, and if so, how? Thanks, Andreas

2 2

[Kamailio-Users] t_newtran/t_reply vs. sl_send_reply
by mayamatakeshi 31 Jul '08

31 Jul '08

Hello, sorry if this is a stupid question, but in the openser.cfg that comes with the source, I see the following: if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } Could you tell me why we use t_newtran/t_reply for cases -1,-3 and sl_send_reply for case -2. I read docs for modules sl and tm but I still don't get it why we shoud created a transaction in one situation and not in the other. Regards, takeshi

2 1

[Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 30 Jul '08

30 Jul '08

I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination... Any help is greatly appreciated...

1 0

[Kamailio-Users] pua_bla and presence
by Gustavo Mistrinelli 30 Jul '08

30 Jul '08

Hey, I have the same problem , using the same openser configuration file, only adding pua_bla "server_address" parameter on 1.3.2 bla stop working. I installed 2 openser once with 1.3.0 and the other with openser 1.3.2, then start openser using full debug log level to see what happens with each one. I'm using Polycom 650 hardphones Basic console error log ------------------------------------- ERROR:pua_bla:bla_handle_notify: content length= 0 WARNING:tm:t_unref: script writer didn't release transaction http://pastebin.com/m4a1b7969 (OpenSER 1.3.2 debug logs) Full debug log ----------------------- http://pastebin.com/m177c219a (OpenSER 1.3.0 full log) http://pastebin.com/m36f3ffca (OpenSER 1.3.2 full log) The main difference that I saw comparing both logs are remarked in pastbin, and refer to: "DBG:pua:print_subs: pres_uri" Please see below both configuration parameters Openser Parameters ---------------------------------- # Presence server module - common API loadmodule "presence.so" modparam("presence", "server_address", "sip:192.168.4.10") modparam("presence", "max_expires", 3600) # Presence server module - presence & watcher info and XCAP loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) # Common API for presence user agent client loadmodule "pua.so" # BLA extension for PUA loadmodule "pua_bla.so" modparam("pua_bla", "default_domain", "192.168.4.10") modparam("pua_bla", "header_name", "Sender") modparam("pua_bla", "server_address", "sip:192.168.4.10") (Only for OpenSER 1.3.2 I'm sending configuration of affected routes (Main, Register, Presence) so you can see if there is an error. route { # Sanity Check Section #..... # Loose Route Section if (loose_route()) { xlog("L_INFO", "$mi Loose Routing"); if(!has_totag()) { xlog("L_INFO", "OPENSER_NAME: $mi Initial loose-routing rejected - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); route(ACC_FORBIDDEN_ROUTE); return; } if (is_method("CANCEL") || is_method("ACK")) { xlog("L_INFO", "$mi CANCEL/ACK"); route(BASE_LOCAL_ROUTE); return; } if (is_method("SUBSCRIBE") && $rd == "OPENSER_IP") { # in-dialog subscribe requests route(PRESENCE_ROUTE); return; } if(is_method("BYE")) { route(ACC_BYE_ROUTE); xlog("L_INFO", "(ACC) $mi Setting account_flag"); setflag(ACC_FLAG); route(BASE_LOCAL_ROUTE); return; } }; # Call Type Processing Section if (method=="INVITE") { route(INVITE_ROUTE); return; }; if (method=="REGISTER") { route(REGISTER_ROUTE); return; } if(is_method("PUBLISH|SUBSCRIBE")) { route(PRESENCE_ROUTE); return; } if(is_method("NOTIFY")) { if($hdr(event)=="dialog;sla") { route(PRESENCE_ROUTE); return; } } } route[REGISTER_ROUTE] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html xlog("L_NOTICE", "OPENSER_NAME: $mi Register Route - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if ((!www_authorize("OPENSER_DOMAIN", "device")))) { xlog("L_INFO", "OPENSER_NAME: $mi Register Authentication Failed - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("OPENSER_DOMAIN", "0"); sl_send_reply("401", "Unauthorized"); exit; } else xlog("L_INFO", "OPENSER_NAME: $mi Register Authentication OK -\ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if ( !(src_ip=="LOCAL_PAIR_IP" && src_port==LOCAL_PAIR_PORT) ) { if($hdr(User-Agent) =~ "PolycomSoundPointIP-SPIP_650-UA") { xlog("L_NOTICE", "$mi Polycom BLA set FLAG \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); bla_set_flag(); } if (!save("location")) { xlog("L_NOTICE", "$mi $fu save(location) Failed\n"); sl_reply_error(); return(0); }; } consume_credentials(); return; } route[PRESENCE_ROUTE] { # ------------------------------------------------------------------- # Presence Route # ------------------------------------------------------------------- xlog("L_INFO","$mi: Presence Route"); if (!t_newtran()){ sl_reply_error(); exit; }; if(is_method("PUBLISH")){ if($hdr(Sender)!= NULL) { handle_publish("$hdr(Sender)"); } else { handle_publish(""); } t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } else if(is_method("NOTIFY")) { if(bla_handle_notify()) { t_reply("200", "OK"); t_release(); } exit; } Ok, I hope that provided information help to see if there is a problem on my configuration or a is a bug on OpenSER 1.3.2 Thanks in advance, Cheers, Gustavo Mistrinelli

1 0

[OpenSER-Users] OpenSER as NAT traversal proxy HELP!
by Joris Dobbelsteen 30 Jul '08

30 Jul '08

Dear, I'm really trying to use OpenSER as a NAT traversal SIP proxy, since my home phone keeps breaking voice channels (the box was not intended behind NAT and I'm, of course, using a configuration that no so well supported). What is the idea: SIP transactions should travel this way: ZyXEL UA <-> SIP Proxy <-> NAT Firewall (iptables) <-> {Internet} RTP should travel this way: ZyXEL UA <-> NAT Firewall & RTPProxy <-> {Internet} My current test is using X-Lite with voipbuster, but that doesn't really work. It seems that registers are functioning, at least X-Lite reports itself being registered. Voice calls always end up in timeouts, so something is really going wrong here, it might be authentication problems? An added problem is that I have just sufficient knowledge of SIP to see what it is doing, without really knowing what to expect exactly. Furthermore I have virtually no knowledge of OpenSER. I've quite a hard time even grasping the configuration I typed in. This is not really helpful What I do know: * SIP Proxy traffic is flowing. * SIP INVITES don't work at all. * SIP to RTP is communication, but I don't know if RTP is actually flowing. I stole most of the configuration from the "04 NAT Traversal" slides of the "Italy 2007 Admin course", to which there is link on the documentation site. I adapted it to make it work with the debian supplied OpenSER 1.1. How do I get this all working? What am I getting wrong? I really really appeciate any help I can get to get it working! - Joris Config is this: # ----------- global configuration parameters ------------------------ debug=4 # 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) advertised_address="82.168.191.xx" advertised_port=5060 port=5060 children=4 fifo="/tmp/openser_fifo" # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "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) # -- nathelper params --- modparam("nathelper", "rtpproxy_sock", "udp:192.168.10.6:22222") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@82.168.191.xx") # ------------------------- 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; }; # NAT detection route(2); # 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("sip.familiedobbelsteen.nl", "subscriber")) { www_challenge("sip.familiedobbelsteen.nl", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # INVITE will have branch flag 6 set after loopup("location") setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; 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 (subst_uri('/(sip:.*);nat=yes/\1/i')) { setflag(6); }; if (isflagset(5) || isflagset(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")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { force_rtp_proxy("", "82.168.191.xx"); t_on_failure("2"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[2] { if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); }; } onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status =~ "(183)|(2[0-9][0-9])") { force_rtp_proxy(); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); }; exit; }

7 17

[OpenSER-Users] reply route failure
by Fero Vlkolinsky 30 Jul '08

30 Jul '08

Hello, I am getting the folowing error on communication with one of our application servers: tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 404 This is in situation where I have two triggerings to the AS (orig and term) and term subscriber is not registered to OpenSER. OpenSER thus returns 404 not found which is then sent back to OpenSER to be forwarded to orig leg of the AS. I had the very same behavior with other AS in our lab and found out that it was because of tag added to To header in AS for the 404 response. So i removed the tag and the call worked fine. However with this AS it doesn't seem to work. Is there a way to hack pick_branch so that I see what are the existing branches and why it does not match any? Or is there any easier way to find out the reason? Thanks a lot, Fero --- reklama ----------------------------------------------------- Hľadáš DOM – BYT – POZEMOK – KANCELÁRIU ? http://www.reality.sk

2 1

[Kamailio-Users] Add a table to openser DB
by Diego Costantini 30 Jul '08

30 Jul '08

Hi, I need to use something like this (openser on a pc with 2 network cards): http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy since I couldn't find any documentation to properly create a "location_external" table, I followed this procedure: http://www.tutorialspoint.com/mysql/mysql-clone-tables.htm Now I have the table I need, but when I start openser I get this error: Jul 30 15:49:32 [8374] ERROR:usrloc:register_udomain: Invalid table version (use openser_mysql.sh reinstall) Jul 30 15:49:32 [8374] ERROR:registrar:domain_fixup: failed to register domain Jul 30 15:49:32 [8374] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 451 Jul 30 15:49:32 [8374] CRITICAL:core:fix_expr: fix_actions error Jul 30 15:49:32 [8374] ERROR:core:main: failed to fix configuration with err code -1 Jul 30 15:49:32 [8374] NOTICE:presence:destroy: destroy module ... How can I make Openser happy to accept my new table? Diego

2 1

[Kamailio-Users] Getting SNMPStats to work
by Alex Massover 30 Jul '08

30 Jul '08

Hello! I'm trying to setup SNMPStats on Debian Sid. I have: dev5:~# cat /etc/openser/openser.cfg | grep snmp loadmodule "snmpstats.so" modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 1234) modparam("snmpstats", "MsgQueueMajorThreshold", 5678) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_minor_threshold", 600) dev5:~# cat /etc/snmp/snmpd.conf | grep agent master agentx agentXSocket tcp:localhost:705 dev5:~# cat /etc/snmp/snmpstats.conf agentXSocket tcp:localhost:705 dev5:~# snmpd -f -D -Dagentx -x tcp:localhost:705 2>&1 | less No log handling enabled - turning on stderr logging registered debug token agentx, 1 netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() agentx_register_app_config_handler: registering .conf token for "agentxsocket" agentx_register_app_config_handler: registering .conf token for "agentxperms" agentx_register_app_config_handler: registering .conf token for "agentxRetries" agentx_register_app_config_handler: registering .conf token for "agentxTimeout" /etc/snmp/snmpd.conf: line 115: Warning: Unknown token: system.sysDescr.0. Turning on AgentX master support. agentx/config: port spec: tcp:localhost:705 agentx/master: initializing... agentx/master: initializing... DONE NET-SNMP version 5.4.1 agentx/master: unexpected callback op 4 agentx/master: handle pdu (req=0x365b5eac,trans=0x0,sess=0x0) agentx/master: open 0x86b37e8 agentx/master: opened 0x86c2028 = 6 with flags = a0 agentx/master: send response, stat 0 (req=0x365b5eac,trans=0x0,sess=0x0) But it doesn't work: dev5:~# snmpget -v2c -c public localhost openserSIPEntityType.0 OPENSER-SIP-COMMON-MIB::openserSIPEntityType.0 = No Such Object available on this agent at this OID Am I missing something? -- Best Regards, Alex Massover

1 0

[Kamailio-Users] OpenSER + Asterisk (Voicemail) Issue....
by Gerard A. Matthew 30 Jul '08

30 Jul '08

I'm having this little issue when implementing the voicemail feature. My openser.cfg looks like this in the failure route: if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemail.mydomain.com:5061"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } On my asterisk end after the time out, i'm viewing the following: SELECT * FROM sipusers WHERE name = 'XXXXXX' i.e XXXXXX = the PSTN number i'm using to call into the IP phone that's connected to OpenSER There seems to be a simple mixup with the number that is sent to asterisk. Obviously there is no user with the PSTN number, however there is one with the called number. Any idea as to what wold be causing this? Have I provided enough information? Regards, Gerard.

2 1

[OpenSER-Users] Polycom Phones produce no wringing when dialing pots lines on the pstn
by Alex G 30 Jul '08

30 Jul '08

I'm sure i'm not the first to experience this. My polycom phones when calling out a pstn gateway recieve far end ringing when dialing cellphones in the US, but when i try to call a pots line i get no ringing, only silence until the end point answers the call. Does anyone have a solution for this? From what i can see in the sip captures, the gateway sends a 180 ringing on the pots line calls, but when i call a cellphone i recieve a 183 Session Progress where the phones do provide the ringing. A polycom to polycom call shows the same 180 ringing, but in this case they do provide ringing. Just to be clear, the ringing i am talking about is the rinigng you here in the earpiece after a dial. Any insight or help would be very appreciated. I have sip captures that i can post, but opted not to incase someone has seen this and knows the answer first. Thanks so much!

3 2

Re: [Serusers] use_dns_cache and call forwarding
by caio 29 Jul '08

29 Jul '08

Hi Miklos, This happens only when I try to do a call to a subcriber having forwarding enabled. Normal calls success ok. I'm not sure about the append_branch() usage and how to work with 'on busy' and 'no answer' forwarding features. Thanks your comments.. Claudio On Thu, Jul 24, 2008 at 4:58 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > Hi, > > On 07/23/2008 05:22 PM, caio wrote: >> >> Hi guys, >> I have a problem with call forwarding (on busy or not answer).., the >> call fall into the failure_route(1) block, and AVP checking and pushto >> does its job. >> >> Here a snipped of code of failure_route(1) (flag 27 means call fwd if >> no answer): >> >> if (isflagset(27) && t_check_status("408")) { # if fwd no >> answer is set and reply msg is 408 >> >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> resetflag(27); >> >> avp_print(); >> >> log(1, "LOG: --> appending new branch..."); >> if (!append_branch()){ >> t_reply("500", "Too many branches?!"); >> drop; >> } >> >> log(1, "LOG: --> Calling route_6..."); >> route(6); >> break; >> >> Route(6) finally calls route(1), where t_relay is called. >> But have a dns failure..and the call never is forwarded. >> >> == ser.log == >> >> DEBUG: mk_proxy: doing DNS lookup... >> get_record: lookup(_sip._udp.wsa.lab, 33) failed_ >> sip_resolvehost: no SRV record found for wsa.lab, trying 'normal' >> lookup..._ >> check_via_address(190.244.33.5, 190.244.33.5, 0)_ >> ERROR: udp_send: sendto(sock,0x2ab32599c720,910,0,0x2ab325999e68,16): >> Operation not permitted(1)_ >> msg_send: ERROR: udp_send failed_ >> ERROR: t_forward_nonack: sending request failed_ >> DEBUG: add_to_tail_of_timer[4]: 0x2ab325999e80_ >> DEBUG: add_to_tail_of_timer[0]: 0x2ab325999ea0_ >> ERROR: w_t_relay (failure mode): forwarding failed_ >> >> A question regarding this trouble.....Can I disable use_dns_cache ? >> SER version is 0.9.7. > > there is no DNS cache in version 0.9.x as I recall, but anyway, your problem > does not seem to be related to DNS, but to the UDP message sending: SER > fails to send out the UDP packet, "Operation not permitted". Is there no > firewall or routing problem at the proxy side? > > Miklos > >> What do you recommend? >> If you need my ser.cfg I can paste it... >> >> Thanks.. >> > -- caio -- caio

2 8

[Kamailio-Users] rtpproxy scalability.
by Alex Balashov 29 Jul '08

29 Jul '08

Does anyone have any reliable data on rtpproxy dimensioning, i.e. just about how many calls per box (and what kind of specs) it can really handle in actually-existing practice, not in theory? The one time I used it for NAT traversal was on a pretty beefy quad-CPU box, and I started seeing clicks and distortions in the calls beyond a load of 90 or so. However, I was never able to decisively ascertain that the cause is not the network. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] new list for announcements
by Henning Westerholt 29 Jul '08

29 Jul '08

The Kamailio project has now a dedicated list for announcements. There will be only important messages posted to this list, like release announcements, security advisories, informations about important changes in the project organisation and community events. The posting frequency will be problably not more then a few per month. All postings will be moderated. The address of the list is: kamailio-announce(a)lists.kamailio.org, to subscribe visit: http://lists.kamailio.org/cgi-bin/mailman/listinfo/kamailio-announce Best regards, Henning

1 0

Re: [Kamailio-Users] Prepaid system using openSER
by Darren Sessions 29 Jul '08

29 Jul '08

I’ve used SER and now OpenSER (uhhm, Kamailio) for the sip proxy and cdr generation (along with media proxy and cdrtool) and Asterisk as the ivr piece for years. Using a proxy like OpenSER/Kamailio to front your IVR servers is your best bet. The combination has worked extremely well for me. - Darren -- www.darrensessions.com On 7/28/08 6:52 PM, "Rana Dhekial" <dhekial(a)msn.com> wrote: > Hi, > > We have a requirement to build a prepaid system using OpenSER. I > took a look at CDRTool which looks quite promising. But I am not > sure how to build the Call control module and provide IVR type > information to the end users. > > If any one has done similar type of job, please respond. > > thanks, > > Rana > > Keep your kids safer online with Windows Live Family Safety. Help > protect your kids. <http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa… > >

1 0

[Kamailio-Users] HELP: Create new table for Nathelper
by Diego Costantini 29 Jul '08

29 Jul '08

Hi, I repropose my unanswered question. I was following this sample configuration for a multihomed server: http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy and it mentions: "For this example to work, you must also follow the nathelper module instructions for creating new location_internal and location_external tables. Be careful to use underscores ("_") and not hyphens ("-") when creating the new tables." But I couldn't find any instruction anywhere. Can someone address me to the correct place please? Thanks, Diego

2 4

[Serusers] What is the behavior during Update of Contact List in ser2?
by Tomasz Zieleniewski 29 Jul '08

29 Jul '08

Hi, I have a question concerning registration. If my UA performs a successful registration and during contact update it sends first REGISTER request with the Authorization header already present with new nonce and response params, will ser respond with 401 or OK. Will functions such us radius_www_authenticate("$fd.digest_realm"); and www_authenticate("$td.digest_realm", "credentials"); return success?? BTW how is the avp fd.digest_realm set? Now when we try to do this ser responds with 401. Is this correct? Kind regards Tomasz

1 0

[Serusers] SER and Radus auth + acc
by Samuel Muller 29 Jul '08

29 Jul '08

Hello all, I've a single question : Does anyone made a SER v2.0 with a working Radius authentication and accounting ? I actually am interest to get some tips, I've many problems about AVPs, DB relations, proxy authentication, ... and i don't really find any full updated docs. The config : SER 2.0, mySQL, Radiator. Many thanks in advance ! Samuel MULLER

1 0

[Serusers] retargeting - rerouting - redirecting
by .:: Francesco la Torre ::. 29 Jul '08

29 Jul '08

Dear all, few days ago I found in a script posted in this mail-list a function that allow to forward a sip message to another host in this way function_name("ip.host.target.message","port"); Is there anyone that know the name of this function ? Thank a lot, Flt

2 3

[Kamailio-Users] Prepaid system using openSER
by Rana Dhekial 29 Jul '08

29 Jul '08

Hi, We have a requirement to build a prepaid system using OpenSER. I took a look at CDRTool which looks quite promising. But I am not sure how to build the Call control module and provide IVR type information to the end users. If any one has done similar type of job, please respond. thanks, Rana _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

2 1

[Kamailio-Users] Pronunciation
by Darren Sessions 29 Jul '08

29 Jul '08

So, what's the phonetic pronunciation of 'kamailio' ? :)

1 0

[Kamailio-Users] Project continuity and near future.
by Adrian Georgescu 28 Jul '08

28 Jul '08

Alex Nothing has changed. The King is dead. Long live the King! http://en.wikipedia.org/wiki/The_King_is_dead._Long_live_the_King%21

1 0

[Kamailio-Users] Project continuity and near future.
by Sergio Gutierrez 28 Jul '08

28 Jul '08

Hello to all the members. As user of the project and responsible of an implementation of it in a production large scale environment. it is a little bit worrying reade terms as "conflicting interests" related to project in the news published today. According to this new change within the project, which I feel go further than renaming it, what could the board members to comment about the RoadMap, philosophy and near future of Kamailio (Formerly known as OpenSER)? One of the biggest warranties that users have felt ourselves is exactly the support and knowledge from the founders and leaders, and it would be very worrying not been able to count with it anymore. Thanks in advance for your attention, and I apologize in advance if this question is out off topic. Best regards. Sergio Gutiérrez.

5 7

[Kamailio-Users] [OpenSER-Users] [MediaProxy] local vs. remote domains?
by Adrian Georgescu 28 Jul '08

28 Jul '08

The readme file that comes with the software explains this. Adrian I've been looking at the mediaproxy (v1) code, and am a little confused why it has the concept of "local" and "remote" domains and why it treats them differently. Can anyone shed light on this? Max.

2 1

[Kamailio-Users] [OpenSER-Users] [MediaProxy] If early media is longer than "idleTimeout" the RTP session is terminated
by Adrian Georgescu 28 Jul '08

28 Jul '08

Hi Iñaki, As far as we are concerned we do not maintain MediaProxy 1.x anymore. So is likely this will not get fixed. All new developments and bug fixing are done against 2.0 Adrian >>>>>>> Hi, I use MediaProxy 1.9.1 with OpenSer. idleTimeout=50 seconds I've received an INVITE and a 183 with early media during 52 seconds and after that the 200 was received. But the RTP session in MediaProxy was terminated after 50 seconds (idleTimeout). Why? Is a still no answered call considered "idle"? This is the syslog:

1 0

Re: [Kamailio-Users] [OpenSER-Users] Firewall
by Henning Westerholt 28 Jul '08

28 Jul '08

On Thursday 24 July 2008, CSB wrote: > I have an error which is driving me crazy: > > Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:core:udp_send: > sendto(sock,0x81aaed8,825,0,0xb61216f0,16): Operation not permitted(1) > > If I turn the firewall off, start a call and then turn the firewall on the > call continues successfully. However the next call is not successful. We > use Mediaproxy and I suspect that is trying to do some communication that > is being blocked by the firewall but that is only a suspicion. The dialogue > for this call is shown at the bottom. Hi CSB, how about debugging your firewall rules step by step and test each rule? This way you should be able to isolate the problematic one. Cheers, Henning

1 0

[OpenSER-Users] OpenSER becomes Kamailio Project
by Adrian Georgescu 28 Jul '08

28 Jul '08

July 28, 2008 OpenSER becomes Kamailio Project OpenSER project and the community around it have grown considerably in the past three years. Because of its extensive development and high adoption rate, conflicting interests related to the OpenSER name and similar trademarks emerged. Consequently, we have been forced to find a new name for the project that stands on itself and represent the project in its future expression. Starting with version 1.4, OpenSER is called Kamailio. Existing OpenSER users can keep on enjoying the benefits of the best open source SIP server available on the market under this new name. All project infrastructure like the website and mailing list will be in the future accessible under the kamailio.org domain, but we will keep the old one functional for a transition period to make the switch as easy as possible for you. Kamailio is a hawaiian word. Kama'ilio means talk, to converse. It was chosen for its special flavour. It is hopefully easy to remember and the meaning fits well with the project purpose. We hope you like it too! For more information visit: http://www.kamailio.org -- http://www.openser.org/mos/view/Management/

1 0

[OpenSER-Users] [MediaProxy] local vs. remote domains?
by Max Bowsher 28 Jul '08

28 Jul '08

I've been looking at the mediaproxy (v1) code, and am a little confused why it has the concept of "local" and "remote" domains and why it treats them differently. Can anyone shed light on this? Max.

1 0

[Serusers] SER 2.0 on Gentoo?
by caio 28 Jul '08

28 Jul '08

Hi all, Did anyone install SER v2.0 on a Gentoo? If yes, is there an ebuild for this? Thanks.. -- caio

1 0

[OpenSER-Users] How to controle minutes per user ?
by Thiago Rondon 28 Jul '08

28 Jul '08

Hello, I'm with a server openser running smoothly. - Users are authenticating. (MySQL) - I succeeded in redirecting the calls to my "supplier" voip. But I can not control the number of minutes that each user can use, that is, I would like to organize users by groups to say: -- Group A users can make connections/calls up to 2 minutes. -- Group B users can make connections/calls up to 3 minutes. Thank you very much! -Thiago

3 2

[OpenSER-Users] openser resends invite on failure -- please help
by Krunal Patel 27 Jul '08

27 Jul '08

Hi Here is the scenario: *UAC UAS(openser) termination server* | | | |-----------------INVITE---------------------->| | |<----------------100 Trying-------------------| | | |-----------------INVITE----------------------> | | |<----------------100 Trying--------------------| | |<----------------403 Forbidden---------------| | |-----------------ACK-------------------------> | |<-----------------403 Forbidden-------------| | |-----------------ACK-------------------------> | | | |-----------------INVITE----------------------> | |<-----------------403 Forbidden-------------| | |-----------------ACK-------------------------> | | | |-----------------INVITE----------------------> | |<-----------------403 Forbidden-------------| | |-----------------ACK-------------------------> | | | |-----------------INVITE----------------------> | |<-----------------403 Forbidden-------------| | |-----------------ACK-------------------------> | | | |-----------------INVITE----------------------> | |<-----------------403 Forbidden-------------| | |-----------------ACK-------------------------> | | This how openser sends re sends invite to termination server indefinitely. Please need your help. It would be appreciated. Thanks, Krunal Patel

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Re: [OpenSER-Users] Res: how to make openser db ready for serMyAdmin
by Andrew Pogrebennyk 27 Jul '08

27 Jul '08

Hi Flavio, Thank you! It turned out to be a non-obvious firewall thing that prohibited communication with DB host. However now I'm unable to login as 'admin' with password 'secret': serMyAdmin sends auth.invalid error cause. In catalina.out log there's only this message: [Fatal Error] :-1:-1: Premature end of file. Admin user was created with http://www.sermyadmin.org/openser/openser.sql *************************** 1. row *************************** id: 6 username: admin domain: setup password: secret first_name: Administrator last_name: email_address: admin(a)setup.com datetime_created: 2008-04-09 16:14:22 ha1: e2df32663fbff3bb2bd22ed7a4a35636 ha1b: 7d9f741c07c122817e51c4fc817b15dd timezone: NULL rpid: NULL version: NULL password_hash: NULL auth_username: NULL class: NULL domain_id: NULL role_id: NULL 1 row in set (0.00 sec) Any ideas? Flavio Goncalves wrote: > Hi Andrew, > In the first time you access the page (http://ip:8080/serMyAdmin) the fields are created. Checl updated installation procedures at www.sermyadmin.org. > Best regards, > Flavio E. Goncalves -- Sincerely, Andrew Pogrebennyk

1 0

[OpenSER-Users] how to make openser db ready for serMyAdmin
by Andrew Pogrebennyk 26 Jul '08

26 Jul '08

Hi, The normal openser database is not appropriate for executing this SQL file: http://www.sermyadmin.org/openser/openser.sql E.g. in `domain` table there's no `version` column, the `jsec_user_role_rel` table doesn't exist at all and the `subscriber` table doesn't have these columns: `version`, `password_hash`, `auth_username`, `class`, `domain_id` and `role_id`. I wonder how to update the structure of openser DB so that it would be ready for serMyAdmin. Thanks in advance. -- Sincerely, Andrew Pogrebennyk

1 0

[Serusers] ser2.0 - logging missed calls
by Davor Jovanovic 26 Jul '08

26 Jul '08

Hi all, I need help with logging of missed calls for offline users. acc_db module is loaded but acc_db_missed function unrecognized. I tired to use this function with ser2.0_rc1 and latest ser-2.0.0+cvs20080709 but the error is same. Or could I solve this in some other way? And if someone could point me to acc_db module documentation because I can't find it on iptel.org site. Thank you /davor ser -V version: ser 2.0.0+cvs20080506 (i386/linux) ... loadmodule "/usr/local/ser_20_06052008/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser_20_06052008/lib/ser/modules/acc_db.so" modparam("acc_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc_db", "log_missed_flag", 4) ... if (lookup_contacts("location")) { . . . } else { acc_db_missed("480 - User temp. unavail.", "missed_calls"); sl_reply("480", "User temp. unavail."); drop(); } ... 0(23273) find_export_record: <acc_db_missed> not found 0(23273) parse error (1116,46-47): unknown command, missing loadmodule? /davor

1 0

[Serusers] Need help to better understand transactions
by Pauba, Kevin L 25 Jul '08

25 Jul '08

Hi, I'm running SER V0.9.6 and I'm having a problem with CANCELs. A client sends and invite to SER that relays it on to a gateway. In this particular scenario, a CANCEL is send by the client after SER responds with the 100 trying. I had thought that SER would match the INVITE based on the branch tag and automatically send a CANCEL on to the gateway. Unfortunately, SER cannot find a matching transaction and, thus, the GATEWAY never receives a CANCEL.. Here's a snippet from the debugging output: ... 0(28802) SIP Request: 0(28802) method: <INVITE> 0(28802) uri: <sip:4027160348@wic.west.com> 0(28802) version: <SIP/2.0> 0(28802) parse_headers: flags=1 0(28802) Found param type 232, <branch> = <z9hG4bK1249933199>; state=16 0(28802) end of header reached, state=5 0(28802) Header Type: 1 0(28802) parse_headers: Via found, flags=1 0(28802) parse_headers: this is the first via 0(28802) header field type 1, name=<Via>, body=<SIP/2.0/UDP 10.27.192.94:5060;branch=z9hG4bK1249933199> 0(28802) preparing to run routing scripts... 0(28802) parse_headers: flags=128 0(28802) Header Type: 8 0(28802) header field type 8, name=<From>, body=<<sip:4027161000@10.27.192.94>;tag=3673666958> 0(28802) end of header reached, state=9 0(28802) DEBUG: get_hdr_field: <To> [31]; uri=[sip:4027160348@wic.west.com] 0(28802) DEBUG: to body [<sip:4027160348@wic.west.com> ] 0(28802) Header Type: 4 0(28802) header field type 4, name=<To>, body=<<sip:4027160348@wic.west.com>> 0(28802) Header Type: 32 0(28802) header field type 32, name=<Call-ID>, body=<514+43+1215800190+24+t1b(a)10.27.192.94> 0(28802) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(28802) Header Type: 16 0(28802) header field type 16, name=<CSeq>, body=<2 INVITE> 0(28802) Header Type: 64 0(28802) header field type 64, name=<Contact>, body=<<sip:e514@10.27.192.94>> 0(28802) Header Type: 33554432 0(28802) header field type 33554432, name=<user-agent>, body=<oSIP/wicsip-3.3.4> 0(28802) Header Type: 1073741824 0(28802) header field type 1073741824, name=<x-west-rg>, body=<0:17> 0(28802) Header Type: 1024 0(28802) header field type 1024, name=<Content-Type>, body=<application/sdp> 0(28802) DEBUG: get_hdr_body : content_length=111 0(28802) Header Type: 2048 0(28802) header field type 2048, name=<Content-Length>, body=<111 > 0(28802) found end of header 0(28802) Header Type: -1 0(28802) DEBUG: is_maxfwd_present: max_forwards header not found! 0(28802) grep_sock_info - checking if host==us: 12==13 && [wic.west.com] == [10.27.193.119] 0(28802) grep_sock_info - checking if port 5060 matches port 5060 0(28802) grep_sock_info - checking if host==us: 12==13 && [wic.west.com] == [10.27.193.119] 0(28802) grep_sock_info - checking if port 5060 matches port 5060 0(28802) check_self: host != me 0(28802) subst_run: running. r=0 0(28802) subst_run: matched (0, 0): [] 0(28802) subst_run: running. r=1 0(28802) subst_f: lst was (nil) 0(28802) subst_run: running. r=0 0(28802) subst_run: matched (0, 36): [sip:9990001254027160348@wic.west.com] 0(28802) textops: subst_uri_f: match - old uri= [sip:9990001254027160348@wic.west.com], new uri= [sip:9990001254027160348@10.27.202.157] 0(28802) DEBUG: t_newtran: msg id=36 , global msg id=35 , T on entrance=(nil) 0(28802) parse_headers: flags=-1 0(28802) parse_headers: flags=60 0(28802) t_lookup_request: start searching: hash=34792, isACK=0 0(28802) DEBUG: RFC3261 transaction matching failed 0(28802) DEBUG: t_lookup_request: no transaction found 0(28802) DEBUG: add_param: tag=3673666958 0(28802) end of header reached, state=29 0(28802) SER: new INVITE 0(28802) parse_headers: flags=-1 0(28802) check_via_address(10.27.192.94, 10.27.192.94, 0) 0(28802) WARNING:vqm_resize: resize(0) called 0(28802) DEBUG: reply sent out. buf=0x810be30: SIP/2.0 1..., shmem=0x422c3c60: SIP/2.0 1 0(28802) DEBUG: _reply_light: finished 0(28802) DEBUG: mk_proxy: doing DNS lookup... 0(28802) check_via_address(10.27.192.94, 10.27.192.94, 0) 0(28802) DEBUG: add_to_tail_of_timer[4]: 0x422c2ac0 0(28802) DEBUG: add_to_tail_of_timer[0]: 0x422c2ad0 0(28802) SER: new transaction fwd'ed 0(28802) DEBUG:destroy_avp_list: destroying list (nil) 0(28802) receive_msg: cleaning up 0(28802) ... 0(28802) SIP Request: 0(28802) method: <CANCEL> 0(28802) uri: <sip:4027160348@wic.west.com> 0(28802) version: <SIP/2.0> 0(28802) parse_headers: flags=1 0(28802) Found param type 232, <branch> = <z9hG4bK1249933199>; state=16 0(28802) end of header reached, state=5 0(28802) Header Type: 1 0(28802) parse_headers: Via found, flags=1 0(28802) parse_headers: this is the first via 0(28802) header field type 1, name=<Via>, body=<SIP/2.0/UDP 10.27.192.94:5060;branch=z9hG4bK1249933199> 0(28802) preparing to run routing scripts... 0(28802) parse_headers: flags=128 0(28802) Header Type: 8 0(28802) header field type 8, name=<From>, body=<<sip:4027161000@10.27.192.94>;tag=3673666958> 0(28802) end of header reached, state=9 0(28802) DEBUG: get_hdr_field: <To> [31]; uri=[sip:4027160348@wic.west.com] 0(28802) DEBUG: to body [<sip:4027160348@wic.west.com> ] 0(28802) Header Type: 4 0(28802) header field type 4, name=<To>, body=<<sip:4027160348@wic.west.com>> 0(28802) Header Type: 32 0(28802) header field type 32, name=<Call-ID>, body=<514+43+1215800190+24+t1b(a)10.27.192.94> 0(28802) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 0(28802) Header Type: 16 0(28802) header field type 16, name=<CSeq>, body=<2 CANCEL> 0(28802) Header Type: 64 0(28802) header field type 64, name=<Contact>, body=<<sip:514+43+1215800190+24+t1b@10.27.192.94>> 0(28802) Header Type: 33554432 0(28802) header field type 33554432, name=<user-agent>, body=<oSIP/wicsip-3.3.4> 0(28802) Header Type: 1073741824 0(28802) header field type 1073741824, name=<x-west-rg>, body=<0:0> 0(28802) DEBUG: get_hdr_body : content_length=0 0(28802) Header Type: 2048 0(28802) header field type 2048, name=<Content-Length>, body=<0> 0(28802) found end of header 0(28802) Header Type: -1 0(28802) DEBUG: is_maxfwd_present: max_forwards header not found! 0(28802) grep_sock_info - checking if host==us: 12==13 && [wic.west.com] == [10.27.193.119] 0(28802) grep_sock_info - checking if port 5060 matches port 5060 0(28802) grep_sock_info - checking if host==us: 12==13 && [wic.west.com] == [10.27.193.119] 0(28802) grep_sock_info - checking if port 5060 matches port 5060 0(28802) check_self: host != me 0(28802) subst_run: running. r=0 0(28802) subst_run: matched (0, 27): [sip:4027160348@wic.west.com] 0(28802) textops: subst_uri_f: match - old uri= [sip:4027160348@wic.west.com], new uri= [sip:4027160348@10.27.202.157] 0(28802) subst_run: running. r=0 0(28802) subst_run: matched (0, 0): [] 0(28802) XLOG: xl_print_log: final buffer length 99 0(28802) time [Fri Jul 11 14:15:52 2008] KLP: CANCEL REQUEST - ruri <sip:9990001254027160348@10.27.202.157> 0(28802) DEBUG: t_check: msg id=40 global id=37 T start=0xffffffff 0(28802) parse_headers: flags=-1 0(28802) parse_headers: flags=60 0(28802) t_lookup_request: start searching: hash=34792, isACK=0 0(28802) DEBUG: RFC3261 transaction matching failed 0(28802) DEBUG: t_lookup_request: no transaction found 0(28802) DEBUG: t_check: msg id=40 global id=40 T end=(nil) 0(28802) XLOG: xl_print_log: final buffer length 64 0(28802) time [Fri Jul 11 14:15:52 2008] KLP: CANCEL REQUEST -- ignoring 0(28802) DEBUG:destroy_avp_list: destroying list (nil) 0(28802) receive_msg: cleaning up 0(28802) I've verified that the branchID of the INVITE matches that of the CANCEL so I had expected the t_lookup_request() to succeed. Do I misunderstand the way transactions work? Any suggestions on how I might get this to work? Any help is appreciated. Thanks!

2 3

[Serusers] Fw: SIP SER-Help me
by vu nt 25 Jul '08

25 Jul '08

----- Forwarded Message ---- From: vu nt <vunt83(a)yahoo.com> To: serusers(a)iptel.org Sent: Tuesday, July 15, 2008 2:05:42 PM Subject: SIP SER-Help me My system is running Ser-0.8.6. The system have problem : tail -f /var/log/messages Jul 15 13:50:35 SIP-1 ser[32724]: ERROR: t_forward_nonack: failure to add branches Jul 15 13:50:36 SIP-1 ser[32727]: ERROR: build_req_buf_from_sip_req: out of memory Jul 15 13:50:36 SIP-1 ser[32727]: ERROR: print_uac_request: no pkg_mem Jul 15 13:50:36 SIP-1 ser[32727]: ERROR: t_forward_nonack: failure to add branches Jul 15 13:50:36 SIP-1 ser[32733]: ERROR: build_req_buf_from_sip_req: out of memory Jul 15 13:50:36 SIP-1 ser[32733]: ERROR: print_uac_request: no pkg_mem Jul 15 13:50:36 SIP-1 ser[32733]: ERROR: t_forward_nonack: failure to add branches Jul 15 13:50:37 SIP-1 ser[32731]: ERROR: build_req_buf_from_sip_req: out of memory Jul 15 13:50:37 SIP-1 ser[32731]: ERROR: print_uac_request: no pkg_mem Jul 15 13:50:37 SIP-1 ser[32731]: ERROR: t_forward_nonack: failure to add branches Jul 15 13:50:39 SIP-1 ser[32731]: ERROR: build_req_buf_from_sip_req: out of memory Jul 15 13:50:39 SIP-1 ser[32731]: ERROR: print_uac_request: no pkg_mem Jul 15 13:50:39 SIP-1 ser[32731]: ERROR: t_forward_nonack: failure to add branches Jul 15 13:50:39 SIP-1 ser[32731]: ERROR: build_req_buf_from_sip_req: out of memory But when I monitor by top command on Linux system, the Ram is not out off. top - 14:02:52 up 7 days, 2:01, 1 user, load average: 0.37, 0.35, 0.30 Tasks: 86 total, 1 running, 85 sleeping, 0 stopped, 0 zombie Cpu(s): 1.3% us, 0.7% sy, 0.0% ni, 91.5% id, 6.3% wa, 0.0% hi, 0.2% si Mem: 1034424k total, 525060k used, 509364k free, 234916k buffers Swap: 2044072k total, 0k used, 2044072k free, 154564k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 32733 root 16 0 38656 22m 34m S 1.0 2.3 1:34.70 ser 32716 root 16 0 38656 23m 34m S 0.7 2.3 1:37.83 ser 32722 root 15 0 38648 22m 34m S 0.7 2.3 1:29.31 ser Can you help me solve this problem ? . Thank you very much BRs Vungo

2 1

[Serusers] LCR module, ser0.9.6
by Deepak 25 Jul '08

25 Jul '08

I am using LCR module and I have two set of gateways (group id 1 and group id 2). I have defined a list of prefixes and the corresponding gateway group id's. All is working fine vis-à-vis the prefixes and the choice of the gateway group. My problem is that if a RURI is supposed to be handled by gateways in group id 1, and all the gateways in that group are busy, the call then gets forwarded to the gateways in group id 2 although the prefix does not match to the one associated with group id 2. Is there any way to disable this so that if all gateways are busy in group id 1, SER does not forward calls to the group id 2? Thanks in advance

1 0

[OpenSER-Users] change From display name and From URI username
by Mouncif Benniane 25 Jul '08

25 Jul '08

I am trying to change/re-write From URI username and From display name (*$fU and **$fn respectively) to avposp values*I tried: $fU = $avp(s:somevariable); avp_write("$fU","$avp(s:somevariable)"); append_hf("From-username: $avp(s:somevariable)\r\n"); but none of them works, any ideas on how to re-write those variables in SIP messages? Thank you* *

2 1

[OpenSER-Users] [MediaProxy] If early media is longer than "idleTimeout" the RTP session is terminated
by Iñaki Baz Castillo 25 Jul '08

25 Jul '08

Hi, I use MediaProxy 1.9.1 with OpenSer. idleTimeout=50 seconds I've received an INVITE and a 183 with early media during 52 seconds and after that the 200 was received. But the RTP session in MediaProxy was terminated after 50 seconds (idleTimeout). Why? Is a still no answered call considered "idle"? This is the syslog: ----------------------------------------------- # INVITE arrives: Jul 25 15:21:17 server-sip proxydispatcher[21258]: request 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667 77.14.6.152:57102:audio,77.14.6.152:41292:image 77.14.0.144 77.14.0.144 remote 192.168.10.2 remote CS2000_NGSS/9.0 info=from:882534800@77.14.0.144:5060,to:887332031@88.99.3.10:5060,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd,totag: Jul 25 15:21:17 server-sip mediaproxy[21253]: request 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667 77.14.6.152:57102:audio,77.14.6.152:41292:image 77.14.0.144 77.14.0.144 remote 192.168.10.2 remote CS2000_NGSS/9.0 info=totag:,to:887332031@88.99.3.10:5060,from:882534800@77.14.0.144:5060,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd Jul 25 15:21:17 server-sip mediaproxy[21253]: session 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: started. listening on 88.99.3.10:61820,61822 Jul 25 15:21:17 server-sip mediaproxy[21253]: execution time: 0.92 ms Jul 25 15:21:17 server-sip proxydispatcher[21258]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '88.99.3.10 61820 61822' # After 50 seconds the session is ended by MediaProxy: Jul 25 15:22:06 server-sip mediaproxy[21253]: session 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Jul 25 15:22:06 server-sip mediaproxy[21253]: session 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: ended (did timeout). # After 2 seconds the 200 OK arrives so media fails. Jul 25 15:22:08 server-sip proxydispatcher[21258]: lookup 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667 192.168.10.2:19084:audio 88.99.1.213 sip.domain.net local unknown unknown Asterisk=20testDesign info=from:882534800@sip.domain.net,to:887332031@88.99.3.10:5060,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd,totag:as6156af6c Jul 25 15:22:08 server-sip mediaproxy[21253]: lookup 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667 192.168.10.2:19084:audio 88.99.1.213 sip.domain.net local unknown unknown Asterisk=20testDesign info=totag:as6156af6c,to:887332031@88.99.3.10:5060,from:882534800@sip.domain.net,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd Jul 25 15:22:08 server-sip mediaproxy[21253]: session 9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: started. listening on 88.99.3.10:61824 Jul 25 15:22:08 server-sip mediaproxy[21253]: execution time: 0.60 ms Jul 25 15:22:08 server-sip proxydispatcher[21258]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '88.99.3.10 61824' Jul 25 15:22:08 server-sip proxydispatcher[21258]: execution time: 1.09 ms ----------------------------------------------- -- Iñaki Baz Castillo

1 0

[OpenSER-Users] Error with parallel forking and TCP
by Zappasodi Daniele 25 Jul '08

25 Jul '08

Hello, I have a problem with parallel forking and TCP when one connection isn't available and the inv_timeout expires. I have two clients registered with the same username and transport=TCP. If I make a call to this number when one of them is not reachable, the INVITE goes to the phone reachable, if the call timeout expires (fr_inv_timer_avp) Openser sends the CANCEL, but doesn't send anything to the caller. The failure_route doesn't hit. There is also a delay between the t_relay and the forwarding of the packet TCP on the net (around 3 seconds). The TCP packet is forwarded to the available connection only after Openser detects the failure for the closed TCP connection (the INVITE goes on the net only after the message ERROR:tm:t_forward_nonack: sending request failed). This is the trace (debug level 3, Openser version 1.3.2): Jun 25 16:30:01 SAM-IP ser[23888]: R1 t_relay R-uri: <sip:10240@10.45.10.63:2054;transport=tcp;line=zdhv6xi4> Jun 25 16:30:01 SAM-IP ser[23888]: Branch route <sip:10240@10.45.10.63:2054;transport=tcp;line=zdhv6xi4> 1 Jun 25 16:30:01 SAM-IP ser[23888]: Branch route <sip:10240@10.45.10.36:5064;transport=tcp> 2 Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (113) No route to host Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcp_send: connect failed Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:tm:msg_send: tcp_send failed Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:tm:t_forward_nonack: sending request failed Jun 25 16:30:04 SAM-IP ser[23890]: Reply Route: provision 180 - Ringing ip <10.45.10.36:33109> From <sip:10242@10.45.8.249> To <sip:10240@10.45.8.249> Jun 25 16:30:04 SAM-IP ser[23890]: ERROR:core:io_wait_loop_sigio_rt: ignoring event 41 on fd 24 (fm->fd=-1, fm->data=0x406e5488) Jun 25 16:30:04 SAM-IP ser[23900]: ERROR:core:io_wait_loop_sigio_rt: ignoring event 41 on fd 25 (fm->fd=-1, fm->data=0x406c5258) Jun 25 16:30:11 SAM-IP ser[23889]: Reply Route: 487 - <Request Terminated> ip <10.45.10.36:33110> From <sip:10242@10.45.8.249> To <sip:10240@10.45.8.249> Jun 25 16:30:11 SAM-IP ser[23890]: ERROR:core:io_wait_loop_sigio_rt: ignoring event 41 on fd 24 (fm->fd=-1, fm->data=0x406d5370) Jun 25 16:30:11 SAM-IP ser[23891]: ERROR:core:tcp_read: error reading: Connection reset by peer Jun 25 16:30:11 SAM-IP ser[23891]: ERROR:core:tcp_read_req: failed to read ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 3

[OpenSER-Users] Need openser.cfg with RLS working
by mahesh.peddi＠info-spectrum.com 25 Jul '08

25 Jul '08

Hi All, I am currently working on OpenSER-1.3.2. I want to exclude the RLS module but when I did getting some errors. Could you please send me the openser.cfg with RLS working. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

2 1

[OpenSER-Users] Regarding watcher Application
by sureshkumar.jaiswal 25 Jul '08

25 Jul '08

Hi All is openseer is supporting watcher application (NDO) ~Suresh..

3 2

[OpenSER-Users] IPv6 Destination unreachable
by Andhika Gannesha Gemilang 25 Jul '08

25 Jul '08

Trace from cilent: ============================================================================================================= 10:26:36.424197 IP6 2004::90:82.sip > 2004::90:81.sip: SIP, length: 354 `....j.@ ............... ....................jx.REGISTER sip:[2004::90:81] SIP/2.0 Via: SIP/2.0/UDP [2004::90:82]:5060;branch=z9hG4bK1206866792 From: <sip:2002@[2004::90]:81>;tag=1806361594 To: <sip:2002@[2004::90]:81> Call-ID: 1308861798@2004::90:82 CSeq: 1 REGISTER Contact: <sip:2002@[2004::90:82]:5060> Max-Forwards: 70 User-Agent: Linphone-1.7.1/eXosip Expires: 200 Content-Length: 0 10:26:36.424605 IP6 2004::90:81.sip > 2004::90:82.sip: SIP, length: 451 `......@ ............... .....................v.SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [2004::90:82]:5060;branch=z9hG4bK1206866792;received=2004:0:0:0:0:0:90:82 From: <sip:2002@[2004::90]:81>;tag=1806361594 To: <sip:2002@[2004::90]:81>;tag=9bf42f041e5ea2673c94775a0faadb8b.8e04 Call-ID: 1308861798@2004::90:82 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="[2004::90:81]", nonce="4886a61885ddea6cfa1c3e538e26b3ceffab04bf" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 17:26:52.618229 IP6 2004::90:82 > 2004::90:81: ICMP6, destination unreachable, unreachable address 2004::90:82, length 507 `.....:@ ............... .................Im....`......@ ............... .......................SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [2004::90:82]:5060;branch=z9hG4bK1206866792;received=2004:0:0:0:0:0:90:82 From: <sip:2002@[2004::90]:81>;tag=1806361594 To: <sip:2002@[2004::90]:81>;tag=9bf42f041e5ea2673c94775a0faadb8b.8e04 Call-ID: 1308861798@2004::90:82 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="[2004::90:81]", nonce="4886a60c2e9e478a8c8591a61c5ca117814ea990" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 ======================================================================================================== after receiving "401" challenge, the client just can't find the server. why would this happen? it works very well on the ipv4. please help me...

2 1

[OpenSER-Users] Using OpenSER Presence & Snom
by Ross Beer 25 Jul '08

25 Jul '08

Hi, I am using the presence_xml module to provide presence across multiple clients. X-Lite, Bria works well however Snom phones produce the following error: ------------- ERROR ----------------namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Lunch</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Busy</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Lunch</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^------------- END ERROR ----------------It looks like a validation issue with the XML, is there anyway you can switch off validation or update the DTD? Thanks, Ross _________________________________________________________________ Play and win great prizes with Live Search and Kung Fu Panda http://clk.atdmt.com/UKM/go/101719966/direct/01/

1 0

[OpenSER-Users] Regarding watcher Application
by Adrian Georgescu 25 Jul '08

25 Jul '08

This is a feature of OpenXCAP, to retrieve the list of active watchers from OpenSER. It does not require any support in OpenSER. Adrian > Hi All > is openseer is supporting watcher application (NDO) > > ~Suresh.. > > what do you mean by watcher application? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [OpenSER-Users] About openser redirect - forward
by Iñaki Baz Castillo 25 Jul '08

25 Jul '08

El Viernes, 20 de Junio de 2008, Valerio Di Marino escribió: > Thanks for your answer. Ok, but next time continue the thread in the maillist ;) > 2008/6/20 Iñaki Baz Castillo ibc(a)aliax.net: > > AFAIK if a UAC is redirected it will not include the already used > > credentials > > in the new request to the new destination (maybe I'm wrong). > > Do you think is possible to force using the pseudovariables ? No, it's not possible. When a UAAC receives a 302 it generates a completely new request with destination the URI of the "Contact" in the 302. > > If you need solving NAT in your OpenSer then you need to forward the > > request > > in it after make fix NAT. > > How ? Can explain better ? If a UAC receives a 302 it will generate a new INVITE. If the UAC has not configured your OpenSer as outbound proxy, the new INVITE it generates will go directly to the SIP provider, so it's not possible your OpenSer makes fix the NAT (fix "Contact" and so). But maybe your SIP provider makes fix the NAT by itself, or you can try STUN in your UAC's. > > If not you could use forwarding or redirect. > > With redirect, openser will send an address and the uer will contact the > new address: is it correct ? (I am not shure). Yes, OpenSer replies a 302 with the new URI in the "Contact" header. > > In OpenSer to get a redirection you must set the RURI ($ru = ...) and > > reply a > > 302 to the user. > > Can you send ma a routine syntax ? > For example, I would redirect to sip.voipstunt.com Something like: $ru = "sip.voipstunt.com"; send_reply("302", "Redirect to SIP provider"); exit; -- Iñaki Baz Castillo

4 6

[OpenSER-Users] Intervention in provisional responses of stateful INVITE processing.
by Alex Balashov 25 Jul '08

25 Jul '08

Greetings, An OpenSER REQUEST-ROUTE that statefully t_relay()s an INVITE request will automatically pass back a provisional "100 Trying" response to the sender. The problem I have is that I have a database operation that occurs in my main REQUEST-ROUTE that can potentially, under certain unfavourable circumstances, be somewhat latent. This operation must run before the relaying can occur, since its results are used to rewrite the RURI, etc. When the operation lags, sometimes it exhausts the T-timers and results in retransmission of the INVITE, or, for equipment configured in a particularly conservative way, even a failure. So, what I need is some way to pass a provisional "100 Trying" at the beginning of the REQUEST-ROUTE immediately in order to placate the sender. I can't just reply statelessly, because I need the stateful processing for other things related to call accounting, so I must use 'tm.' And if I send a stateless "100 Trying" reply, the subsequent t_relay() will not know that and duplicate it, and although as far as I know this does not violate the RFC, it is still a situation I would like to avoid in principle. What I need to know is if there is some way in 'tm' to disable automatic provisional responses in a temporary or ephemeral fashion, or else to disable them entirely and then re-enable them later. Thank you so much in advance for all of your help! -- 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

2 2

[OpenSER-Users] Why 'sip_trace' desn't show ACK for a "200 OK" from OpenSer to UAS?
by Iñaki Baz Castillo 24 Jul '08

24 Jul '08

(Sorry, I sent this mail to devel maillist) Hi, the first I do in my OpenSer script is: -------------------------------------------------------- modparam("siptrace", "trace_flag", FLAG_SIPTRACE) modparam("siptrace", "trace_on", 1) route { ### SipTrace. if ! is_method("OPTIONS|REGISTER") { sip_trace(); setflag(FLAG_SIPTRACE); } ... -------------------------------------------------------- When an INVITE gets a 200 OK the UAC sends an ACK (end to end). This ACK from UAC to OpenSer is stored in 'sip_trace' table, but the same ACK from OpenSer to the UAS (in-dialog ACK) is not stored, why? Is there any reason for this? AFAIK this ACK is like any in-dialog request and 'sip_trace' stores correctly both legs of any in-dialog request, except in the case of an ACK for a 200 OK. Since the "special" treatment OpenSer performs in the ACK, maybe there is a technical issue that makes imposible ACK from OpenSer to UAS to appear in "sip_trace"? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 2

[OpenSER-Users] Identifying call branches for multiple contacts.
by Alex Balashov 24 Jul '08

24 Jul '08

Greetings, If I have multiple contacts registered concurrently for one binding with the usrloc/registrar modules, is there a way I can get a count for, or iteratively traverse somehow, the call branches created when a request goes out to that binding? In other words, if I have UAC A and UAC B sharing a binding, is there some way I can access the distinct call paths programmatically within the route plan? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 1

[OpenSER-Users] integrating OpenSER + RTPproxy for recording the conservation
by LetMeKnow 24 Jul '08

24 Jul '08

Hello user's OpenSER and RTPproxy Am to integrate the openser and rtprpxy for live recording of converstions. Am just to try to how to do the recording in openser and rtpprpxy Am using openser-1.3.2 and Rtpproxy-1.2 . in my Lab, means proxy and UAC's all are in internal LAN. In openser.cfg if ( method=="INVITE") { force_rtp_proxy(); start_recording(); t_on_reply("1"); t_relay(); exit; } onroute_reply[1] { if (status=="200") force_rtp_proxy(); } when i observer in Rtp Proxy in debug mode. I got this : received command "16034_2 U 26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg. 192.168.4.6229334 f0007919;1" new session 26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg., tag f0007919;1 requested, type strong hash_table_append: hash(26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg.) = 168 new session on a port 35004 created, tag f0007919;1 pre-filling caller's address with 192.168.4.62:29334 sending reply "16034_2 35004 192.168.4.102 " received command "16034_3 L 26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg. 192.168.4.2382650 f0007919;1 1f668e3e;1" lookup on ports 35004/35006, session timer restarted pre-filling callee's address with 192.168.4.238:2650 sending reply "16034_3 35006 192.168.4.102 " received command "16034_4 R 26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg. 1f668e3e f0007919" record request failed: session 26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg., tags 1f668e3e/f0007919 not found sending reply "16034_4 E8 " received command "16034_5 D 26685862e1635844MGYwNDNmODM0MjM0YTIwNWY3Mzc2NzZiZjQwYTJhNzg. f0007919 1f668e3e" forcefully deleting session 1 on ports 35004/35006 RTP stats: 0 in from callee, 6 in from caller, 6 relayed, 0 dropped RTCP stats: 2 in from callee, 2 in from caller, 4 relayed, 0 dropped session on ports 35004/35006 is cleaned up sending reply "16034_5 0 " Audio is fine on both UAC side. problem in its starting and when i use start_recording. Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead 91-9999882776

2 2

Re: [OpenSER-Users] openser + mysql clusters
by Stanisław Pitucha 24 Jul '08

24 Jul '08

----- "Sajith T S" <sajith(a)gmail.com> wrote: > How is the overall experience like re. deploying openser with mysql > clusters? Good enough. Just setup A record pointing at 2 or more api nodes and set them up with virtual ips (heartbeat works quite well...). Openser will recover if one of the nodes fail. > Are there gotchas etc that need to be taken care of? (For > example, a 2006 article [1] says that "The MySQL NDB engine currently > runs its database completely in memory. This means that you have to > be > able to fit your database in memory." But this is not documented as > a > limitation in mysql faq.) It's a feature and it's in the first sentence of cluster overview: "... enables clustering of in-memory databases" ;) Gotchas: - don't bother with 5.0 - it's got strange issues - 5.1.23 was the last version of 5.1.X with ndb. Now you have to compile carrier grade edition from source. - ping database once in a while (every database in a cluster) by *writing* to some table. ndb sometimes just locks itself, but allows selects (locks internally - I've never used lock statements). Restarting api node is enough in that case. - No schema change needed - it just works with standard tables. - Inserts / updates will be slower - not much, but anyway. - hang out in #mysql-ndb channel on freenode -> you'll learn some magic ndb-mgm commands which are sometimes not documented. Otherwise, it's not that different from a standard mysql database. --- Stan

6 8

[OpenSER-Users] Presence & XML DTD Namespace Error
by Ross Beer 24 Jul '08

24 Jul '08

Hi, I am using the presence_xml module to provide presence across multiple clients. X-Lite, Bria works well however Snom phones produce the following error: ------------- ERROR ---------------- namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Lunch</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Busy</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Lunch</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^ ------------- END ERROR ---------------- It looks like a validation issue with the XML, is there anyway you can switch off validation or update the DTD? Thanks, Ross _________________________________________________________________ Invite your Facebook friends to chat on Messenger http://clk.atdmt.com/UKM/go/101719649/direct/01/

1 0

[OpenSER-Users] Firewall causes ERROR:tm:msg_send: udp_send failed
by CSB 24 Jul '08

24 Jul '08

I have an error which is driving me crazy: Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:core:udp_send: sendto(sock,0x81aaed8,825,0,0xb61216f0,16): Operation not permitted(1) Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:tm:msg_send: udp_send failed This is firewall related because when I turn iptables off the problem goes away. But although all ACCEPT and DENY messages are logged by the firewall, there is no corresponding message logged. When a call is made between two UACs and the far end attempts to answer the call, the call is not answered successfully. The dialogue is shown below. If I turn the firewall off, start a call and then turn the firewall on the call continues successfully. However the next call is not successful. We use Mediaproxy and I suspect that is trying to do some communication that is being blocked by the firewall but that is only a suspicion. The dialogue for this call is shown at the bottom. When comparing these two dialogues with the firewall turned on: - when the callee answers, OpenSER receives a 200 OK but doesn't immediately pass that on to the caller - after a number of 200 messages from the callee, OpenSER sends the 200 to the caller but the Contact is the private IP address whereas when the firewall is turned off it is the public IP address I have also included the iptables config. Note that some lines are commented out due to the testing I'm doing and IP addresses have been changed. Any advice on the changes required would be appreciated. Thanks **************************** SIP dialogue with firewall on U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2222 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.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D;rport=5060;received=58.28.001.0 01..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35982 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.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 192.168.1.124:5065ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=566CA8D1-4C8E0458>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK4b4d0d4aF803AD55..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************************** SIP dialogue with Firewall off U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2224 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.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366;rport=5060;received=58.28.001.00 1..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 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.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=3D7BF99A-F3B3ACE5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.2..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************** iptables config $IPTABLES -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A OUTPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT # # Rule 0 (tun0,tun1,tun2) # echo "Rule 0 (tun0,tun1,tun2)" # # # $IPTABLES -N In_RULE_0 $IPTABLES -A INPUT -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A In_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A In_RULE_0 -j ACCEPT $IPTABLES -N Out_RULE_0 $IPTABLES -A OUTPUT -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A Out_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A Out_RULE_0 -j ACCEPT $IPTABLES -A Out_RULE_0 -j ACCEPT # # Rule 1 (lo) # echo "Rule 1 (lo)" # # # $IPTABLES -N In_RULE_1 $IPTABLES -A INPUT -i lo -p icmp -m icmp --icmp-type any -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j In_RULE_1 #$IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_1 $IPTABLES -A In_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A In_RULE_1 -j ACCEPT $IPTABLES -N Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p icmp -m icmp --icmp-type any -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j Out_RULE_1 #$IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_1 $IPTABLES -A Out_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A Out_RULE_1 -j ACCEPT # # Rule 2 (eth0) # echo "Rule 2 (eth0)" # # # $IPTABLES -N Out_RULE_2 $IPTABLES -A OUTPUT -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A FORWARD -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A Out_RULE_2 -j LOG --log-level warning --log-prefix "RULE 2 -- ACCEPT " $IPTABLES -A Out_RULE_2 -j ACCEPT # # Rule 3 (eth0) # echo "Rule 3 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E561900.0 $IPTABLES -A INPUT -i eth0 -d 147.202.001.001 -m state --state NEW -j Cid485B0E561900.0 $IPTABLES -N Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E561900.1 $IPTABLES -N In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 203.89.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 58.28.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 64.38.001.001 -j In_RULE_3 $IPTABLES -A In_RULE_3 -j LOG --log-level warning --log-prefix "RULE 3 -- ACCEPT " $IPTABLES -A In_RULE_3 -j ACCEPT # echo "Rule 4 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E6A1900.0 $IPTABLES -A OUTPUT -o eth0 -s 147.202.001.001 -m state --state NEW -j Cid485B0E6A1900.0 $IPTABLES -N Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E6A1900.1 $IPTABLES -N Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 203.89.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 58.28.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 64.38.001.001 -j Out_RULE_4 $IPTABLES -A Out_RULE_4 -j LOG --log-level warning --log-prefix "RULE 4 -- ACCEPT " $IPTABLES -A Out_RULE_4 -j ACCEPT # # Rule 5 (eth0) # echo "Rule 5 (eth0)" # # # $IPTABLES -N In_RULE_5 $IPTABLES -A INPUT -i eth0 -p tcp -m tcp -m multiport -d 147.202.001.001 --dports 5060,22,443,80,53,25060,8008 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 10000:20000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 35000:36000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport -d 147.202.001.001 --dports 1194,5065,5060,53,10000:20000,35000:36000 -m state --state NEW -j In_RULE_5 $IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_5 $IPTABLES -A In_RULE_5 -j LOG --log-level warning --log-prefix "RULE 5 -- ACCEPT " $IPTABLES -A In_RULE_5 -j ACCEPT # # Rule 6 (eth0) # echo "Rule 6 (eth0)" # # # $IPTABLES -N Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p tcp -m tcp -m multiport -s 147.202.001.001 --dports 22,53,80,443,5060,8008,25060,25 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 10000:20000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 35000:36000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport -s 147.202.001.001 --dports 53,1194,5060,5065,1813,123,10000:20000,35000:36000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A Out_RULE_6 -j LOG --log-level warning --log-prefix "RULE 6 -- ACCEPT " $IPTABLES -A Out_RULE_6 -j ACCEPT # # Rule 7 (global) # echo "Rule 7 (global)" # # # $IPTABLES -N RULE_7 $IPTABLES -A OUTPUT -j RULE_7 $IPTABLES -A INPUT -j RULE_7 $IPTABLES -A FORWARD -j RULE_7 $IPTABLES -A RULE_7 -j LOG --log-level warning --log-prefix "RULE 7 -- DENY " $IPTABLES -A RULE_7 -j DROP # #

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[Serusers] use_dns_cache and call forwarding
by caio 24 Jul '08

24 Jul '08

Hi guys, I have a problem with call forwarding (on busy or not answer).., the call fall into the failure_route(1) block, and AVP checking and pushto does its job. Here a snipped of code of failure_route(1) (flag 27 means call fwd if no answer): if (isflagset(27) && t_check_status("408")) { # if fwd no answer is set and reply msg is 408 if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); avp_print(); log(1, "LOG: --> appending new branch..."); if (!append_branch()){ t_reply("500", "Too many branches?!"); drop; } log(1, "LOG: --> Calling route_6..."); route(6); break; Route(6) finally calls route(1), where t_relay is called. But have a dns failure..and the call never is forwarded. == ser.log == DEBUG: mk_proxy: doing DNS lookup... get_record: lookup(_sip._udp.wsa.lab, 33) failed_ sip_resolvehost: no SRV record found for wsa.lab, trying 'normal' lookup..._ check_via_address(190.244.33.5, 190.244.33.5, 0)_ ERROR: udp_send: sendto(sock,0x2ab32599c720,910,0,0x2ab325999e68,16): Operation not permitted(1)_ msg_send: ERROR: udp_send failed_ ERROR: t_forward_nonack: sending request failed_ DEBUG: add_to_tail_of_timer[4]: 0x2ab325999e80_ DEBUG: add_to_tail_of_timer[0]: 0x2ab325999ea0_ ERROR: w_t_relay (failure mode): forwarding failed_ A question regarding this trouble.....Can I disable use_dns_cache ? SER version is 0.9.7. What do you recommend? If you need my ser.cfg I can paste it... Thanks.. -- caio

2 1

Re: [OpenSER-Users] CDRTool with OpenSER
by Rana Dhekial 24 Jul '08

24 Jul '08

Hi Dan, Please find output of 'dkpg -l 'php4-*'. Looks like php4-mysql is installed. openser:~# dpkg -l 'php4-*'Desired=Unknown/Install/Remove/Purge/Hold| Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed|/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err: uppercase=bad)||/ Name Version Description+++-==============-==============-============================================ii php4-cgi 4.4.4-8+etch6 server-side, HTML-embedded scripting languagun php4-cli <none> (no description available)ii php4-common 4.4.4-8+etch6 Common files for packages built from the phpun php4-curl <none> (no description available)ii php4-dev 4.4.4-8+etch6 Files for PHP4 module developmentun php4-memcache <none> (no description available)ii php4-mysql 4.4.4-8+etch6 MySQL module for php4ii php4-pear 4.4.4-8+etch6 PHP Extension and Application Repository (trun php4-pgsql <none> (no description available)/etc/php4/apache2/php.ini has the following lines.. extension=mysql.so Please let me know any additional config required. thanks, Rana Date: Thu, 24 Jul 2008 07:02:42 +0200From: danb(a)sms4sip.comTo: dhekial(a)msn.comSubject: Re: [OpenSER-Users] CDRTool with OpenSERCC: users(a)lists.openser.org Hi Rana,you need to have php4-mysql installed and loaded in your php code.DanB On Thu, Jul 24, 2008 at 3:00 AM, Rana Dhekial <dhekial(a)msn.com> wrote: We are trying to install cdrtool on the same machine where OpenSER ( 1.2.2 ) is running. We get the following error message when trying to access the page: http://<ipaddress>/CDRToolFatal error: Call to undefined function: mysql_pconnect() in /var/www/CDRTool/phplib/db_mysql.inc on line 73MySQL version is 5.0.32, Apache version is 2.2.3, php version is 4.please let us know if there is anything we can do to fix this problem. thanks, Rana Use video conversation to talk face-to-face with Windows Live Messenger. Get started._______________________________________________Users mailing listUsers@lists.openser.orghttp://lists.openser.org/cgi-bin/mailman/listinfo/users _________________________________________________________________ Time for vacation? WIN what you need- enter now! http://www.gowindowslive.com/summergiveaway/?ocid=tag_jlyhm

2 1

[OpenSER-Users] CDRTool with OpenSER
by Rana Dhekial 24 Jul '08

24 Jul '08

We are trying to install cdrtool on the same machine where OpenSER ( 1.2.2 ) is running. We get the following error message when trying to access the page: http://<ipaddress>/CDRToolFatal error: Call to undefined function: mysql_pconnect() in /var/www/CDRTool/phplib/db_mysql.inc on line 73MySQL version is 5.0.32, Apache version is 2.2.3, php version is 4.please let us know if there is anything we can do to fix this problem. thanks, Rana _________________________________________________________________ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_W…

2 1

[OpenSER-Users] debian build service for trunk
by Henning Westerholt 23 Jul '08

23 Jul '08

Hi all, i've just setup a new Debian repository on the devel.openser.org machine. It replaces the old offering of daily debian packages from the trunk for several debian releases [1]. This new repository provides a more convinient method for keeping track of the bleeding edge of the OpenSER development with standard Debian system administration tools like apt-get. I extend the supported distributions too, now also packages for debian testing (i386 and amd64) are build. With the help of this service its really easy for you to test new features in the trunk and provide feedback *hint, hint*. ;-) Just add some lines [2] (depending on your system version) to your "/etc/apt/sources.list" file, do an "apt-get update" and you can install (or update to) the fresh code direct from the trunk. :-) I also provided a repository key, to stop the security complains from apt.. [3] Please keep in mind that this packages are only provided for testing, and _not_ for use in production environments! The update cycle is daily, just as for the old service. Let me know if you encounter any issues. Have fun, Henning [1] http://devel.openser.org/packages/ [2] Debian stable: deb http://devel.openser.org/debian etch main deb-src http://devel.openser.org/debian etch main Debian testing, also usable for (K)ubuntu deb http://devel.openser.org/debian lenny main deb-src http://devel.openser.org/debian lenny main Debian unstable deb http://devel.openser.org/debian sid main deb-src http://devel.openser.org/debian sid main [3] apt gpg key - add the key to your apt key ring like this: "wget -O - http://devel.openser.org/gpg_key.asc | apt-key add - " -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

2 1

Re: [OpenSER-Users] Problem with OpenXCAP
by Schumann Sebastian 23 Jul '08

23 Jul '08

OK, good it works now. Sebastian ________________________________ From: remi.druilhe(a)orange-ftgroup.com [mailto:remi.druilhe@orange-ftgroup.com] Sent: Wednesday, 23. July 2008 13:41 To: Schumann Sebastian Subject: Re: [OpenSER-Users] Problem with OpenXCAP You were right, it was the proxy. I had forgotten to export HTTP_PROXY in my environment variable of user root (as I start OpenXCAP as root). Now it works and I can continue to configure it. Thank Remi Schumann Sebastian a écrit : Yes, indeed, this might be a problem. The server needs to get the XML scheme from the internet. As you cannot pass proxy settings to OpenXCAP (at least afaik) there might be the problem. It cannot connect and get the file for validation (what is also told by the error message). Try it without proxy - if possible - just to eliminate that potential cause. Sebastian ________________________________ From: remi.druilhe(a)orange-ftgroup.com [mailto:remi.druilhe@orange-ftgroup.com] Sent: Wednesday, 23. July 2008 11:51 To: Schumann Sebastian Subject: Re: [OpenSER-Users] Problem with OpenXCAP Hi Sebastian, Thank for your answer. I think the problem doesn't come from the internet connection because my server is connected to it but behind a proxy, maybe a problem ? Regards Remi Schumann Sebastian a écrit : Dear Remi I had problems once there as well and mine were caused by the lack of internet connection. Maybe this is also the reason for you?? Seems all requirements for install are there as it shows it's starting. Check http://www.openxcap.org/wiki/Running if you find some useful hints maybe there. Best regards Sebastian -----Original Message----- From: remi.druilhe(a)orange-ftgroup.com [mailto:remi.druilhe@orange-ftgroup.com] Sent: Wednesday, 23. July 2008 10:41 To: users(a)lists.openser.org Subject: [OpenSER-Users] Problem with OpenXCAP Hi, I am trying to install and configure OpenXCAP 0.9.9 on Ubuntu 8.04.1 but I encounter a problem after starting the service. In fact, when I look at the log file, OpenXCAP tell me this : Log opened. Log opened. Starting Open XCAP 0.9.9 Starting Open XCAP 0.9.9 and after some minutes : Traceback (most recent call last): Traceback (most recent call last): File "/usr/bin/openxcap", line 56, in <module> File "/usr/bin/openxcap", line 56, in <module> from xcap.server import XCAPServer from xcap.server import XCAPServer File "/usr/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> File "/usr/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> from xcap import authentication from xcap import authentication File "/usr/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> File "/usr/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> from xcap.appusage import getApplicationForURI from xcap.appusage import getApplicationForURI File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 494, in <module> File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 494, in <module> 'resource-lists': ResourceListsApplication(open(os.path.join(schemas_directory, 'resource-lists.xsd'), 'r').read(), Storage()), 'resource-lists': ResourceListsApplication(open(os.path.join(schemas_directory, 'resource-lists.xsd'), 'r').read(), Storage()), File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 64, in __init__ File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema etree.XMLSchemaParseError: Document is not valid XML Schema I know I am not the first to have the problem and I searched in the mailing list and on Google to solve it. I tried some solutions without success. Do you have any idea that could help me ? Thanks -- Rémi Druilhe -- zze-DRUILHE Remi RD-CORE-LAN LAN/CORE/M2V Tel : Email: remi.druilhe(a)orange-ftgroup.com <mailto:remi.druilhe@orange-ftgroup.com> R&D Lannion 2, avenue Pierre Marzin 22307 LANNION Cedex - France Mailer ThunderBird sous YourDev Linux Installez simplement votre poste de travail Linux avec http://web2000.rd.francetelecom.fr -- zze-DRUILHE Remi RD-CORE-LAN LAN/CORE/M2V Tel : Email: remi.druilhe(a)orange-ftgroup.com <mailto:remi.druilhe@orange-ftgroup.com> R&D Lannion 2, avenue Pierre Marzin 22307 LANNION Cedex - France Mailer ThunderBird sous YourDev Linux Installez simplement votre poste de travail Linux avec http://web2000.rd.francetelecom.fr

1 0

Re: [OpenSER-Users] Problem with OpenXCAP
by Schumann Sebastian 23 Jul '08

23 Jul '08

Yes, indeed, this might be a problem. The server needs to get the XML scheme from the internet. As you cannot pass proxy settings to OpenXCAP (at least afaik) there might be the problem. It cannot connect and get the file for validation (what is also told by the error message). Try it without proxy - if possible - just to eliminate that potential cause. Sebastian ________________________________ From: remi.druilhe(a)orange-ftgroup.com [mailto:remi.druilhe@orange-ftgroup.com] Sent: Wednesday, 23. July 2008 11:51 To: Schumann Sebastian Subject: Re: [OpenSER-Users] Problem with OpenXCAP Hi Sebastian, Thank for your answer. I think the problem doesn't come from the internet connection because my server is connected to it but behind a proxy, maybe a problem ? Regards Remi Schumann Sebastian a écrit : Dear Remi I had problems once there as well and mine were caused by the lack of internet connection. Maybe this is also the reason for you?? Seems all requirements for install are there as it shows it's starting. Check http://www.openxcap.org/wiki/Running if you find some useful hints maybe there. Best regards Sebastian -----Original Message----- From: remi.druilhe(a)orange-ftgroup.com [mailto:remi.druilhe@orange-ftgroup.com] Sent: Wednesday, 23. July 2008 10:41 To: users(a)lists.openser.org Subject: [OpenSER-Users] Problem with OpenXCAP Hi, I am trying to install and configure OpenXCAP 0.9.9 on Ubuntu 8.04.1 but I encounter a problem after starting the service. In fact, when I look at the log file, OpenXCAP tell me this : Log opened. Log opened. Starting Open XCAP 0.9.9 Starting Open XCAP 0.9.9 and after some minutes : Traceback (most recent call last): Traceback (most recent call last): File "/usr/bin/openxcap", line 56, in <module> File "/usr/bin/openxcap", line 56, in <module> from xcap.server import XCAPServer from xcap.server import XCAPServer File "/usr/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> File "/usr/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> from xcap import authentication from xcap import authentication File "/usr/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> File "/usr/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> from xcap.appusage import getApplicationForURI from xcap.appusage import getApplicationForURI File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 494, in <module> File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 494, in <module> 'resource-lists': ResourceListsApplication(open(os.path.join(schemas_directory, 'resource-lists.xsd'), 'r').read(), Storage()), 'resource-lists': ResourceListsApplication(open(os.path.join(schemas_directory, 'resource-lists.xsd'), 'r').read(), Storage()), File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 64, in __init__ File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema etree.XMLSchemaParseError: Document is not valid XML Schema I know I am not the first to have the problem and I searched in the mailing list and on Google to solve it. I tried some solutions without success. Do you have any idea that could help me ? Thanks -- Rémi Druilhe -- zze-DRUILHE Remi RD-CORE-LAN LAN/CORE/M2V Tel : Email: remi.druilhe(a)orange-ftgroup.com <mailto:remi.druilhe@orange-ftgroup.com> R&D Lannion 2, avenue Pierre Marzin 22307 LANNION Cedex - France Mailer ThunderBird sous YourDev Linux Installez simplement votre poste de travail Linux avec http://web2000.rd.francetelecom.fr

1 0

[OpenSER-Users] Problem with OpenXCAP
by remi.druilhe＠orange-ftgroup.com 23 Jul '08

23 Jul '08

Hi, I am trying to install and configure OpenXCAP 0.9.9 on Ubuntu 8.04.1 but I encounter a problem after starting the service. In fact, when I look at the log file, OpenXCAP tell me this : Log opened. Log opened. Starting Open XCAP 0.9.9 Starting Open XCAP 0.9.9 and after some minutes : Traceback (most recent call last): Traceback (most recent call last): File "/usr/bin/openxcap", line 56, in <module> File "/usr/bin/openxcap", line 56, in <module> from xcap.server import XCAPServer from xcap.server import XCAPServer File "/usr/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> File "/usr/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> from xcap import authentication from xcap import authentication File "/usr/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> File "/usr/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> from xcap.appusage import getApplicationForURI from xcap.appusage import getApplicationForURI File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 494, in <module> File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 494, in <module> 'resource-lists': ResourceListsApplication(open(os.path.join(schemas_directory, 'resource-lists.xsd'), 'r').read(), Storage()), 'resource-lists': ResourceListsApplication(open(os.path.join(schemas_directory, 'resource-lists.xsd'), 'r').read(), Storage()), File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 64, in __init__ File "/usr/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema etree.XMLSchemaParseError: Document is not valid XML Schema I know I am not the first to have the problem and I searched in the mailing list and on Google to solve it. I tried some solutions without success. Do you have any idea that could help me ? Thanks -- Rémi Druilhe

2 1

[OpenSER-Users] How to get a substring?
by Daniel Cardoso Alves 22 Jul '08

22 Jul '08

Hi... I am using the openser 1.1.0 and I would like get a substring of FROM... For example, I have the code in my script: ---------------------------------------------------------- avp_write("$fU","$avp(s:777)"); avp_print(); ---------------------------------------------------------- And it shows in the log: ---------------------------------------------------------- 5(6249) INFO: name=<777> 5(6249) INFO: val_str=<4830353131 / 10> ---------------------------------------------------------- Is possible I get only the first and the second char (in this case is 48) of $fU??? Are there some way? Thanks..... Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

2 1

[OpenSER-Users] PRESENCE module not working and openser.pid file not found
by Megha Saini 22 Jul '08

22 Jul '08

Hi, I am using OpenSER since a few months, but only with basic functionality, i.e. no Presence, Message-waiting functionality, XCAP etc. I configured the server with MySQL on the same machine and I am able to add new users using the openserctl script as: /usr/local/sbin/openserctl add user password user@domain Now, I wish to include Presence functionality to see the status of users on client side. Thus, I loaded the module presence.so and also presence_xml.so. I also uncommenetd other features in openser.cfg that are required for presence funcionality: # ----- presence params ----- modparam("presence|presence_xml", "db_url", "mysql://user:pwd@domain/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:ip:port") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'megs') #modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "max_expires", 3600) modparam("presence", "fallback2db",1) #------------------------------------------------------ However, now if I try to start openser using the command : /usr/local/sbin/openser start, I see that all modules are loaded and my connection "user:pwd@domain/openser" is in the pool. It also displays that all modules have successfully connected to the database. After this when I try to run: /usr/local/lib/openser/openserctl/openserctl start It displays: database engine 'MYSQL' loaded control engine 'FIFO' loaded INFO: Starting OpenSER: ERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed There is no file openser.pid on the path "/var/run/". But previously, I was able to start openser successfully. I also tried starting openser by: /etc/init.d/openser start But it just says: Starting Openser: [FAILED] Can anyone please tell me how to solve this bug. Where can I find openser.pid file? If not this, then please tell me if I am doing some configuration mistake. Another thing, I see that when I am running my client, I am still able to Register to the server and I am gettin 200OK for REGISTER message. I find it strange! I can see that I have all the tables required for "presence" in the openser database, i.e. presentity, active_watchers etc. Also, when I am using SUBSCRIBE-NOTIFY messages on client side, I get 202 Accepted from the server when the other user is online and 404 NOT FOUND when the user is offline. But I see that all tables in 'openser' database are empty except subscriber table in which I can easily make entries using 'openserctl' script file. I find it strange that how my server is keeping the information of the online users when all databse tables are empty. Ideally, it should keep an entry of availabe users in "location" table, but that table is also empty even though I have written in openser.cfg file: modparam("usrloc","db_mode",2) to enable DB persistency for location entries. Please anyone help me clear the doubts, as I need to complete the presence module soon due to strich deadlines. I have been workin on it since 2 weeks already! :( Do I need to have XCAP server necessarily to implement simple presence status using SUBSCRIBE-NOTIFY messages? Thanks in advance, Megs

2 2

[Serusers] Sintax information
by orante.tucceri 22 Jul '08

22 Jul '08

Hi, its possible to write a return () without arguments to interrupt a sub-route and back to the main route? thanks Orante

2 1

[OpenSER-Users] remove clear password from DB
by Ahmed Huraimel 22 Jul '08

22 Jul '08

Hi all, i am currently using openser 1.3.2 with MySQL DB and serMyAdmin as user portal. i configured my openSER to use the hashed passwords while authenticating the users. my issue is that i do not want to store the user password in clear in my DB. can anyone tell me how to do it? i just want to store the hashed password. thanks in advance. Ahmed ALALI

1 0

Re: [OpenSER-Users] Help regarding FIFO error
by Henning Westerholt 22 Jul '08

22 Jul '08

On Tuesday 22 July 2008, asim shahzad wrote: > [..] > Jul 22 06:14:54 professor-laptop /usr/sbin/openser[7129]: > INFO:usrloc:ul_init_locks: locks array size 512 Jul 22 06:14:54 > professor-laptop /usr/sbin/openser[7129]: INFO:registrar:mod_init: ^^^ Nice machine name ;-) > Jul 22 06:14:54 professor-laptop /usr/sbin/openser[7129]: > rc_read_config: can't open /usr/local/etc/radiusclient-ng/radiusclient.conf: > No such file or directory Its a known problem with the acc module for this version, look here: http://www.openser.org/dokuwiki/doku.php/troubleshooting:faq#qerroraccinit_… > it will be good if you can suggest me some good tutorial to start with > thanks For the basics you could read the OpenSER book [1]. You should also take a look at the core cookbook in the wiki and read the module documentation of some basic modules (sl, tm, usrloc, registrar..). In the wiki and the homepage you find also some links to different documentation resources, if you don't want to buy the book/ for more detailed informations. Cheers, Henning [1] http://www.openser.org/mos/view/News/NewsItem/OpenSER-First-Book/

1 0

[OpenSER-Users] Help regarding FIFO error
by asim shahzad 22 Jul '08

22 Jul '08

Hi, i am student and working on openser in order to implement it for a project. but i did the default configuration as given on to the site but i got this error. i checked the module that is loaded or not its loaded but not working so please help me regarding that how can i solve this error ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. Regards Asim _________________________________________________________________ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_W…

2 1

[OpenSER-Users] SoftPhone keep sending REGISTER message
by Andhika Gannesha Gemilang 22 Jul '08

22 Jul '08

Hi, im using Linphone, when im about to connect using ipv6 address, i found that my softphone keep sending REGISTER message; ------------------------------------------------- 11:54:49.475240 IP6 2004:fa22:337:4::82.sip > 2004:fa22:337:4::81.sip: SIP, length: 400 `......@ ..".7.......... ..".7.................aREGISTER sip:[2004:fa22:337:4::81] SIP/2.0 Via: SIP/2.0/UDP [2004:fa22:337:4::82]:5060;branch=z9hG4bK679317669 From: <sip:2002@[2004:fa22:337:4:]:81>;tag=618703948 To: <sip:2002@[2004:fa22:337:4:]:81> Call-ID: 1861704303@2004:fa22:337:4::82 CSeq: 1 REGISTER Contact: <sip:2002@[2004:fa22:337:4::82]:5060> Max-Forwards: 70 User-Agent: Linphone-1.7.1/eXosip Expires: 900 Content-Length: 0 11:54:49.475635 IP6 2004:fa22:337:4::81.sip > 2004:fa22:337:4::82.sip: SIP, length: 437 `......@ ..".7.......... ..".7..................SIP/2.0 200 OK Via: SIP/2.0/UDP [2004:fa22:337:4::82]:5060;branch=z9hG4bK679317669;received=2004:FA22:337:4:0:0:0:82 From: <sip:2002@[2004:fa22:337:4:]:81>;tag=618703948 To: <sip:2002@[2004:fa22:337:4:]:81>;tag=9bf42f041e5ea2673c94775a0faadb8b.a53a Call-ID: 1861704303@2004:fa22:337:4::82 CSeq: 1 REGISTER Contact: <sip:2002@[2004:fa22:337:4::82]:5060>;expires=892 Server: OpenSER (1.3.0-notls (i386/linux)) Content-Length: 0 ------------------------------------------------------------- the softphone continue to sending REGISTER even after receiving 200 OK... until the server send 400 Bad Request; ---------------------------------------------------------- 11:58:29.577922 IP6 2004:fa22:337:4::81.sip > 2004:fa22:337:4::82.sip: SIP, length: 486 `......@ ..".7.......... ..".7..................SIP/2.0 400 Bad Request Via: SIP/2.0/UDP [2004:fa22:337:4::82]:5060;branch=z9hG4bK728798624;received=2004:FA22:337:4:0:0:0:82 From: <sip:2002@[2004:fa22:337:4:]:81>;tag=618703948 To: <sip:2002@[2004:fa22:337:4:]:81>;tag=9bf42f041e5ea2673c94775a0faadb8b.3e2a Call-ID: 1861704303@2004:fa22:337:4::82 CSeq: 2 REGISTER Contact: <sip:2002@[2004:fa22:337:4::82]:5060>;expires=873 P-Registrar-Error: Invalid CSeq number Server: OpenSER (1.3.0-notls (i386/linux)) Content-Length: 0 ------------------------------------------------------- PS: the softphone works fine on ipv4 any suggestion would be very helpful...

1 0

[OpenSER-Users] openSER -> Asterisk .... no hangup
by Gerard A. Matthew 21 Jul '08

21 Jul '08

I have a situation where i have: Asterisk => PSTN gateway openSER => registrar/proxy PhoneA -> openSER -> Asterisk -> PSTN phoneB my phones connect to openser... calls are sent out to the PSTN using Asterisk...but when phoneA hangs up, Asterisk does not receive the hang up signal. If PSTN phone B hangs up, asterisk receives the hangup signal. I'm reading forums but not receiving any real solution? is there anyway of fixing this? - Gerard

2 1

[OpenSER-Users] Snom 320 Dialog Event
by Jon Farmer 21 Jul '08

21 Jul '08

I am trying to get presence working on a Snom 320 with OpenSER v 1.3.2.However the phone is not setting the sla parameter in the dialog Event which is required by OpenSER. However the parameter is set on this example from the Snom Wiki http://wiki.snom.com/SIP_Traces Any ideas how to enable it? TIA Regards Jon

2 1

[OpenSER-Users] pua_bla and presence
by Pablo Saro 21 Jul '08

21 Jul '08

Hi there, I was using pua_bla and presence modules successfully with OpenSER 1.3.0 I've upgraded to OpenSER 1.3.2 and now BLA/Presence functionality is broken. Now I'm getting the error: ERROR:pua_bla:bla_handle_notify: content length= 0 Is it possible that something change regarding BLA/Presence in 1.3.2? Nothing in the change log... Thanks in advance. Pablo Saro psaro at google dot com

2 1

[OpenSER-Users] when openser starts - mi_fifo reports "pkg_malloc cannot allocate any more memory"
by Christian Koch 21 Jul '08

21 Jul '08

Hello all, we have installed Openser with mysql on a RedHat Enterprise 4 machine. Details are as follows: Linux Capricadb3 2.6.9-55.ELsmp #1 SMP Fri Apr 20 17:03:35 EDT 2007 i686 i686 i386 GNU/Linux version: openser 1.3.2-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 3590 2008-01-28 17:46:56Z bogdan_iancu $ main.c compiled on 11:26:34 Jul 14 2008 with gcc 3.4.6 When we start openser the child process disappear directly again and the log file shows the following (snippet, this message continues until openser stopped): Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10344]: INFO:nathelper:rtpp_test: rtp proxy <unix:/home/sys/bin/rtpproxy.sock> found, support for it enabled Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10344]: DBG:core:init_mod_child: type=CHILD, rank=2, module=mi_fifo Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10344]: ERROR:mi_fifo:mi_writer_init: pkg_malloc cannot allocate any more memory! Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10344]: CRITICAL:mi_fifo:mi_child_init: failed to init the reply writer Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10344]: ERROR:core:init_mod_child: failed to initializing module mi_fifo, rank 2 Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10344]: ERROR:core:main_loop: init_child failed for UDP listener Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10296]: DBG:core:openser_fork: forking new process "UDP receiver" Jul 14 11:30:10 Capricadb3 /usr/local/sbin/openser[10345]: DBG:mysql:db_mysql_new_connection: connection type is capricadb6 via TCP/IP the end of the log file shows this mem status: Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: Memory status (pkg): Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: fm_status (0x815f100): Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: heap size= 67108864 Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: used= 36512, used+overhead=57200, free=67051664 Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: max used (+overhead)= 63272 Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: dumping free list: Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 1 fragments no.: 6, unused: 0 bucket size: 8 - 8 (first 8) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 2 fragments no.: 22, unused: 0 bucket size: 16 - 16 (first 16) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 3 fragments no.: 3, unused: 0 bucket size: 24 - 24 (first 24) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 4 fragments no.: 1, unused: 0 bucket size: 32 - 32 (first 32) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 5 fragments no.: 3, unused: 0 bucket size: 40 - 40 (first 40) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 16 fragments no.: 4, unused: 0 bucket size: 128 - 128 (first 128) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 35 fragments no.: 1, unused: 0 bucket size: 280 - 280 (first 280) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 39 fragments no.: 1, unused: 0 bucket size: 312 - 312 (first 312) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 66 fragments no.: 1, unused: 0 bucket size: 528 - 528 (first 528) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 67 fragments no.: 1, unused: 0 bucket size: 536 - 536 (first 536) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 76 fragments no.: 1, unused: 0 bucket size: 608 - 608 (first 608) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 119 fragments no.: 1, unused: 0 bucket size: 952 - 952 (first 952) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 513 fragments no.: 1, unused: 0 bucket size: 4104 - 4104 (first 4104) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 680 fragments no.: 1, unused: 0 bucket size: 5440 - 5440 (first 0) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 2060 fragments no.: 1, unused: 0 bucket size: 33554432 - 67108864 (first 0) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: TOTAL: 50 free fragments = 8600 free bytes Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: ----------------------------- Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: Memory status (shm): Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: fm_status (0xa7ff2000): Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: heap size= 268435456 Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: used= 16, used+overhead=17672, free=268417784 Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: max used (+overhead)= 1629112 Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: dumping free list: Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 1 fragments no.: 54, unused: 0 bucket size: 8 - 8 (first 8) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 2 fragments no.: 7, unused: 0 bucket size: 16 - 16 (first 16) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 3 fragments no.: 6, unused: 0 bucket size: 24 - 24 (first 24) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 4 fragments no.: 49, unused: 0 bucket size: 32 - 32 (first 32) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 5 fragments no.: 4, unused: 0 bucket size: 40 - 40 (first 40) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 8 fragments no.: 2, unused: 0 bucket size: 64 - 64 (first 64) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 12 fragments no.: 1, unused: 0 bucket size: 96 - 96 (first 96) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 64 fragments no.: 1, unused: 0 bucket size: 512 - 512 (first 512) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 130 fragments no.: 1, unused: 0 bucket size: 1040 - 1040 (first 1040) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 257 fragments no.: 1, unused: 0 bucket size: 2056 - 2056 (first 2056) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 298 fragments no.: 1, unused: 0 bucket size: 2384 - 2384 (first 2384) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 512 fragments no.: 2, unused: 0 bucket size: 4096 - 4096 (first 4096) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 1280 fragments no.: 2, unused: 0 bucket size: 10240 - 10240 (first 10240) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 2055 fragments no.: 1, unused: 0 bucket size: 1048576 - 2097152 (first 1572864) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: hash = 2062 fragments no.: 1, unused: 0 bucket size: 134217728 - 268435456 (first 266806344) Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: TOTAL: 135 free fragments = 268417784 free bytes Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: ----------------------------- Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: DBG:core:shm_mem_destroy: Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: DBG:core:shm_mem_destroy: destroying the shared memory lock Jul 14 11:30:15 Capricadb3 /usr/local/sbin/openser[10296]: DBG:core:handle_sigs: terminating due to SIGCHLD (END) We have configured openser to start 4 childs in the config file. When we configure 2 childs only it will start. We have tried pretty much everything we can think of, we also used the "DBG_QM_MALLOC" switch in the Makefile.defs but the output in the logfile does not point out anymore helpful infos. We also increased the shared and private memory in the config.h file step by step up to pkg= 64 and shared=256MB without success. We have the same config running on another machine with pretty much the same spec, just a slower CPU and less memory and there it runs with default settings... Any help is very appreciated as we are running out of ideas now. Thanks in advance and best regards, Christian

2 5

[OpenSER-Users] configuring share memory for Openser
by mark morreny 21 Jul '08

21 Jul '08

Hi, I am working on performance test on Openser, and I am getting: ERROR:tm:sip_msg_cloner: no more share memory ERROR:tm:store_reply: failed to alloc' clone memory The share memory allocated is 512M but the error still occurs when concurrent call reached > 100. Is this normal? Is there any benchmarking number that can help to determine the amound of shared memory needed by Openser? Thanks, Mark

2 1

[OpenSER-Users] SIP response code in case of UAS internet down
by Chahn Kim 21 Jul '08

21 Jul '08

Hi, all We are implementing a failure route upon t_relay() to UASs. This failure route is intended to a Caller who's destination UAS is not responding at all, till the request gets timed out, which can be arguably interpreted as UAS' internet is down. The purpose is to re-route incoming calls to any alternatives; i.e. cell pnone or other landline # when the remote user site lost netwrok connectivity. For the scenario above, t_check_status("408") would be the best for determining? Any other better methods or test functions in Openser? Greatly appreciate experienced' guidance. Thanks in advance. John Chahn Kim

2 1

[OpenSER-Users] SRTP and Media proxy
by Peter P GMX 20 Jul '08

20 Jul '08

Just a short question: I want to use OpenSER + mediaproxy with TLS/SRTP. Is mediaproxy able to handle the srtp stream? Best regards Peter

4 6

[OpenSER-Users] Mediaproxy 2.0 Installation
by john brown 20 Jul '08

20 Jul '08

Hello, I'm trying to test Mediaproxy 2.0 however the documentation seems vague in the following - Can I test this on Debian Etch kernel version 2.6.18 ? - The documentation says it cannot be used in combination with any version of OpenSER older than 1.4. The current stable version of OpenSER available is 1.3.2. do I have to run Mediaproxy 2.0 on an unstable version of OpenSER ? - Also it reads on the documentation: To display the history of the media streams CDRTool 6.5 or higher is required but only 6.4.1 is available on the download site. - What is the procedure to install Mediaproxy 2.0 once you download it ? I'm not familiar with Python and the documentation mainly talks about the requiremnts but there is no explanation on how to install it. I copied config.ini.sample to config.ini and edited config.ini, copied debian/mediaproxy-dispatcher.init to /etc/init.d/mediaproxy-dispatcher. Then I tried running media-dispatcher but I got the following error: host.locahosts# /etc/init.d/mediaproxy-dispatcher start Starting MediaProxy dispatcher: media-dispatcherTraceback (most recent call last): File "/usr/bin/media-dispatcher", line 12, in ? from application.process import process, ProcessError ImportError: No module named application.process already running. Thanks John

1 0

Re: [OpenSER-Users] openser and asterisk in the same pc
by Ricky Gutierrez 20 Jul '08

20 Jul '08

hi , thank you for the answer Daniel, but I am a little confused if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (has_totag()) { if (loose_route()) { if(method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; ### section for asterisk #### route[4] { rewritehostport("192.168.10.1:5070"); t_relay(); route(1); } route[10] { append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:[29][0-9]{7}@") { if (is_user_in("credentials","local")) { route(4); exit; } else { sl_send_reply("403", "Not permissions local call"); exit; }; }; if (uri=~"^sip:1[2-9][1-9]{3}@") { if (is_user_in("credentials","int")) { route(4); exit; } else { sl_send_reply("403", "Not permissions internal call"); exit; }; }; if (uri=~"^sip:011[0-9]*@") { if (is_user_in("credentials","international")) { route(4); exit; ettc etc , inside the sip.conf [openser] type=friend context=netsoluciones insecure=very host=localhost disallow=all allow=alaw allow=g729 ;( which the best form is to authenticate the invites or not? if asterisk and openser are in the same box where my error can be ? regards rickygm ----- Original Message ---- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Friday, July 18, 2008 8:16:31 AM Subject: Re: [OpenSER-Users] openser and asterisk in the same pc Hello, >the authentication is required by openser -- that means there is a >mistake in your configuration file if you don't want to authenticate >such INVITE. The phone should respond with a new INVITE having >credentials header -- you have to set the username and password in you >phone as well.

1 0

[OpenSER-Users] Aliases for outgoing calls
by mayamatakeshi 19 Jul '08

19 Jul '08

Hello, I'd like to permit our subscribers to have several aliases for outgoing calls (basically, several number in the PSTN). I think they could set this in the header From of the INVITE to us and we would identify them, authenticate them and finally relay that same alias (and not the subscriber name) to the GW (currently we permit them to have a single alias for outgoing using a field in the subscriber table). I can see the module alias_db can be used to manage aliases, but I understand it can only be used for incoming calls as it provides just a single function alias_db_lookup that operates on the Request-URI. Any ideas? regards, takeshi

4 5

[OpenSER-Users] avp_pushto
by Pablo Saro 18 Jul '08

18 Jul '08

Hi everyone, I have a doubt about the function avp_pushto(). When I do the following: ... avp_pushto("$ru", "$avp(s:devices)/g"); ... If $ru is an array with more than 1 position, avp_pushto works fine, but if both $ru and $avp(s:devices) have only 1 position, the function avp_pushto replace $ru with the new value. It cause loosing of the previous $ru. I believe that avp_pushto function shouldn't replace a value. Any idea? I'm using OpenSER 1.3.2 Thanks in advance. Pablo Saro psaro at google dot com

2 2

[Serusers] Forwarding calls to a ShoreTel System
by Freese, Bill 18 Jul '08

18 Jul '08

I will apologize in advance if this is the wrong forum or if this is an embarrassing newbie question. I am currently using an Trixbox/Asterisk system to connect a Rolm PBX to a ShoreTel phone system via SIP. The asterisk box doesn't correctly support the REFER method, so I'm hoping to use SER to proxy my SIP connections to allow the Rolm users to function correctly with transfers and conferencing. I have installed and configured my SER system using the getting started documentation. I have successfully interconnected my SER system with my Asterisk and with a Cisco 2801 and can place and route calls as I would expect. When I try to place a call through SER to my ShoreTel system, I get the standard "SIP/2.0 404 Not Found". If I compare captures of my successful SIP connection from my Asterisk environment to a capture of the SER attempt, the only real difference I see is that the INVITE TO: field has the IP of my SER system instead of the ShoreTel system. The actual INVITE uri is identifying the ShoreTel device, but the TO field has the SER address. The Asterisk connection has the address of the ShoreTel device in both places. I have tried to look through the docs and searched online, but I did not find any resources on this question. I would appreciate any feedback that could point me in the right direction. Here is a ser -V from my SER system version: ser 0.9.7-pre1 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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 @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 14:05:21 Jul 16 2008 with gcc 4.1.2 I route my calls to the ShoreTel system through the basic PSTN route from the getting started sample: route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("my.IP.addr.here"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); t_on_failure("1"); route(1); } Thank you for any feedback in advance.

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[OpenSER-Users] proxydispatcher execution time
by Ariadne Ramos 18 Jul '08

18 Jul '08

Hello I have installed openser 1.2.3 and mediaproxy 1.9.0 on my server. Im having problems with mediaproxy dispatcher time response. Is taking too much time to answer. I have the following message in the logs proxydispatcher[9601]: execution time: 2183.73 ms Could you please help me on this issue? Thanks in advanced Ariadne L. Ramos Solís Depto. Ingeniería Aprovisionamiento y Señalización VoIP Galaxy Communications Corp. Tel. 2000117 e-mail aramos(a)clarocom.com

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[OpenSER-Users] openser and asterisk in the same pc
by Ricky Gutierrez 18 Jul '08

18 Jul '08

hi list , I am trying to connect openser with asterisk, but when I make calls to the pstn, it shows me a 407 Proxy Authentication Required , am trying to make groups of calls with openser that some extensions have permission of making calls to the pstn, others that alone they call internally and international. I have created 3 groups local, int, international, the extensions that are of the local group have permission I believe me, according to the theory to be able to make local calls,for this I am using asterisk, but this he throws me the message 407. The extension 112 with the address ip 192.168.10.26 belongs to the local group, and he makes an intent of calling to the pstn through asterisk, but this he throws me the message 407 I don't know because thank you for your help +0.399494 192.168.10.26:5060 -> 192.168.10.1:5060 INVITE sip:2642006@192.168.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.26:5060;branch=z9hG4bKb7bfb742a0e47ab4 From: "Ventas" <sip:112@192.168.10.1;user=phone>;tag=ca46913c384c257c To: <sip:2642006@192.168.10.1;user=phone> Contact: <sip:112@192.168.10.26:5060;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: fb973f9069a09ad7(a)192.168.10.26 CSeq: 59632 INVITE User-Agent: Grandstream GXP2000 1.1.5.15 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 382 v=0 o=112 8000 8000 IN IP4 192.168.10.26 s=SIP Call c=IN IP4 192.168.10.26 t=0 0 m=audio 5004 RTP/AVP 0 18 3 2 99 9 4 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/16000 a=rtpmap:4 G723/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000205 192.168.10.1:5060 -> 192.168.10.26:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.26:5060;branch=z9hG4bKb7bfb742a0e47ab4 From: "Ventas" <sip:112@192.168.10.1;user=phone>;tag=ca46913c384c257c To: <sip:2642006@192.168.10.1;user=phone>;tag=d7e793d797690be5ce5420203095040f.2cd4 Call-ID: fb973f9069a09ad7(a)192.168.10.26 CSeq: 59632 INVITE Proxy-Authenticate: Digest realm="192.168.10.1", nonce="48802341954eb0c26cd5eb2dd85f375325fb4a7d", qop="auth" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0

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[OpenSER-Users] How does Operser parse source port from request?
by KevinKinnan 18 Jul '08

18 Jul '08

I wonder that how openser parse the source port from the request. Does the parser function get the port from the header of packet(TCP/UDP) or from the content(by parsing Via header or something) of request message. Regards, Kevin _________________________________________________________________ 这里好多好玩的视频，用鼠标点到视频看看，有惊喜！ http://cnweb.search.live.com/video/results.aspx?q=%E5%A5%A5%E8%BF%90%E5%9C%…

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Re: [OpenSER-Users] Problem getting redirect reason in uac_redirect
by Ruchir 18 Jul '08

18 Jul '08

But after forward, new invite will be generated so avp will be lost, isn't it? On Fri, Jul 18, 2008 at 7:00 PM, kokoska rokoska <kokoska.rokoska(a)post.cz> wrote: > > > > Ruchir napsal(a): > >> This is for explicit call forwarding. I'm testing call forwarding from >> linksys pap2 call forward features which generates "302 moved temporarily" >> message. >> >> > I too :-) > May be I don't understand what you are looking for... > > From my point of view it is simple: > 1. If IS present Diversion header in 302 reply, than push it to avp > 2. If Diversion header IS NOT present, than use "unknow" reason - like in > SS7... > > Best regards, > > kokoska.rokoska > > > On Fri, Jul 18, 2008 at 6:20 PM, kokoska rokoska <kokoska.rokoska(a)post.cz<mailto: >> kokoska.rokoska(a)post.cz>> wrote: >> >> >> >> >> Ruchir napsal(a): >> > I'm using uac_redirect module to handle redirect and accounting from >> > openser. Redirect works fine so as cdr but I'm not getting how to >> write >> > redirect reason in cdr. I didn't find any way to find and store >> redirect >> > reason(call forward, busy, no answer) in CDR. Does anyone know >> how to do it? >> > >> > >> >> >> I'm using something like this (shortened): >> >> in config: >> >> modparam("acc", >> "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") >> >> >> in routing script: >> >> $avp(s:acc_state) = "cfu"; >> avp_printf("$avp(i:901)", "$avp(s:caller_uuid)|$avp(s:acc_state)"); >> >> >> Works very well :-) >> >> Hope this helps, best regards, >> >> kokoska.rokoska >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org <mailto:Users@lists.openser.org> >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> >> >

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[OpenSER-Users] Regarding Wathcer Application
by sureshkumar.jaiswal 18 Jul '08

18 Jul '08

i'm want to implement watcher application in openser .. but did find any idea and what fuctionality wathcer have. can any one help me regarding how i got help regarding this. ~suresh

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[OpenSER-Users] openser + mysql clusters
by Sajith T S 18 Jul '08

18 Jul '08

Hi! How is the overall experience like re. deploying openser with mysql clusters? Are there gotchas etc that need to be taken care of? (For example, a 2006 article [1] says that "The MySQL NDB engine currently runs its database completely in memory. This means that you have to be able to fit your database in memory." But this is not documented as a limitation in mysql faq.) Non clustered experiments were occasionally catastrophic, so can't risk taking that route again. Somebody suggested using drbd [2], but I did not quite like it -- it needs kernel modules, has more administrative overhead, is tied to linux, it is not clear how consistency (ie, state of in-memory data and on-disk data) is taken care of, and in any case ndbcluster appears to be the clustering solution with official stamp of approval. (Related - is postgres any better? Apparently postgres' clustering capabilities are "better", but then cdrtool etc seems to be written with mysql as the primary target.) Thanks, Sajith. [1] http://www.oreillynet.com/pub/a/databases/2006/02/16/ha_mysql_cluster.html [2] http://www.drbd.org/ -- "the lyf so short, the craft so long to lerne." -- Chaucer.

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[OpenSER-Users] How to do =~ case insensitive?
by Iñaki Baz Castillo 18 Jul '08

18 Jul '08

Hi, I want to do a case insensitive match, for example for Privacy header that can contain "id", "ID", "Id"... and other values. I need something as: if ($hdr(Privacy)=~"id"/i) { but this obviously fails. Which is the way to do it? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Openser and mysql script
by Yann Derichard 18 Jul '08

18 Jul '08

Hi, I am using Ubuntu 8.04 and I have just installed Openser 1.3.1 via debian package (with mysql and presence modules) to do presence with my computer but I am not able to find "openser_mysql.sh" script to configure my database. So I downloaded the src package on openser ftp and unpack it but the script is still missing. Is someone could tell me where I could download it ? Or if I did something wrong during my installation ? Regards -- Yann Derichard

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[OpenSER-Users] Limiting maximum number of concurrent calls
by Rana Dhekial 18 Jul '08

18 Jul '08

Hi, We are running openeser version 1.2.2. Currently we have defined 3 groups of subscribers. We have a requirement of limiting max number of concurrent calls per group. For example, 10 maximum number of concurrent ( incoming /outgoing ) calls for subscribers belonging to group 1. How do we go about it. Any sample config will be appreciated. thanks, dhekial _________________________________________________________________ With Windows Live for mobile, your contacts travel with you. http://www.windowslive.com/mobile/overview.html?ocid=TXT_TAGLM_WL_mobile_07…

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[OpenSER-Users] OpenSER sending BYE to private IP behind NAT
by Bruno Machado 17 Jul '08

17 Jul '08

Hi Lee You are not running the 'fix_nated_contact' function for INVITE messages. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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[OpenSER-Users] OpenSER sending BYE to private IP behind NAT
by George Lee 17 Jul '08

17 Jul '08

Hi Experts, I have one SIP UA on private IP behind NAT and another 1 on public IP. They both registered to OpenSER and they could call each other. The problem is on call release. The BYE message sent from the UA on public IP could never get to the other UA on private IP. OpenSER kept on sending BYE to the private IP instead of sending it to the external NAT IP. I tried to use fix_nat_register() and fix_nat_contact() when REGISTER was received from the UA behind NAT but it didn't work. I also tried putting fix_nat_contact() on BYE and still no help. Am I missing anything? modparam("nathelper", "received_avp", "$avp(s:rcv)") . . . . . if (method == REGISTER) { if (nat_uac_test("2") || nat_uac_test("16")) { fix_nated_register(); # or fix_nated_contact(); } } . . . . . if (method == BYE) { fix_nated_contact(); } Thanks, whglee

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[OpenSER-Users] Saving real IP:port using fix_nated_register()
by George Lee 17 Jul '08

17 Jul '08

Hi Experts, In my openser.cfg file, I tried to use the fix_nated_register() function to save the IP:port information when a REGISTER message is received. After I restart OpenSER, none of my SIP UAs could register to it anymore. Network traces showed "port unreachable". Can someone help? Thanks, whglee

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Re: [OpenSER-Users] Does OPENSER support AAAA record (IPv6)?
by Fangyu Ling 17 Jul '08

17 Jul '08

Hi Klaus: Thanks for your answer. After I add dns_try_ipv6=yes in openser.cfg, my problem has been solved. But I wonder why the default value of dns_try_ipv6 is "yes" in version 1.2.x, but "no" in version 1.3.x? For my personal opinion, setting default value to "yes" is better than "no". Best Regards, Fang > Hi! > > Openser does support AAAA, when enabled: > http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#dns_try_ipv6 > > regards > klaus > > Fangyu Ling wrote: >> In my DNS server, there are only AAAA records for any nodes. >> My test topology is as following: >> >> -----+---------+--------+------ >> | | | >> siphone1 openser siphone2 >> (UA1) (UA2) >> >> For the testing procedure, I only use domain name of the URI. >> When UA1 invited UA2 via openser, openser only ask A record >> of UA2. There are no A records in my DNS server! So, openser >> then replied 476 Unresolvable Destination message to UA1. >> >> In normal conditon, if there were no such name of A record, >> it should ask AAAA record instead. >> But why OPENSER only ask A record? >> >> Best Regards, >> Fang

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[OpenSER-Users] Using t_replicate to replicate towards a UA
by Pascal Maugeri 17 Jul '08

17 Jul '08

Hi Is it possible to replicate incoming SIP requests (eg. REGISTER) to another remote user SIP URI instead of host URI: For instance I want to do (having previously registered to OpenSER the user pascal.maugeri(a)company.com): t_replicate("sip:pascal@company.com <sip%3Apascal(a)company.com>"); instead of t_replicate("sip:10.2.3.4:5060"); I tried it with OpenSER 1.3 but I get the error message when it tries to replicate: 3(8113) Main routing block. Method [REGISTER] 3(8113) ERROR: mk_proxy: could not resolve hostname: "company.com" 3(8113) ERROR: uri2proxy: bad host name in URI <sip:pascal@company.net<sip%3Apascal(a)company.net> > 3(8113) ERROR:tm:t_forward_nonack: failure to add branches I'm just wondering if what I want to do is possible ... Regards, Pascal

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[OpenSER-Users] O'Reilly Media is looking for contributors
by Kristian Kielhofner 17 Jul '08

17 Jul '08

Hello OpenSER Users, O'Reilly Media has been working on an Asterisk Cookbook. Using the same printed format as other O'Reilly Cookbooks (Perl, Python, etc) O'Reilly Cookbooks offer practical solutions to problems in a fun "recipe" format. I myself have been working quite a bit with OpenSER in the last couple of years and I've asked the other authors if we should open up submissions for other recipes (including OpenSER). Everyone thinks it's a great idea. We are looking for a potentially new name and URL but the current "Asterisk Cookbook" Wiki can be found here: http://www.asteriskcookbook.com/ We'll be covering "Asterisk and friends". That means Asterisk, OpenSER, Cisco gateways, etc. Pretty much anything VoIP with at least an Open Source slant. Request an account and add your recipes to the recipe heap (you'll see my OpenSER recipes towards the bottom). See you there! -- Kristian Kielhofner http://blog.krisk.org

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[OpenSER-Users] Problem with accounting in case of call forwarding from client
by Ruchir 16 Jul '08

16 Jul '08

I have problem with CDR manipulation in case of call forwarding from client. I'm testing with Linksys pap2 and I've set unconditional call forwarding. When linksys receives the call, it'll send 302, temporarily moved. I have done proper ACL checking using Diversion header in the new INVITE but I'm having difficulty associating the forwarded leg CDR to original call as this is totally new INVITE and none of the fields match in CDR. Can anyone guide me in right direction?

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[OpenSER-Users] Does OPENSER support AAAA record (IPv6)?
by Fangyu Ling 16 Jul '08

16 Jul '08

In my DNS server, there are only AAAA records for any nodes. My test topology is as following: -----+---------+--------+------ | | | siphone1 openser siphone2 (UA1) (UA2) For the testing procedure, I only use domain name of the URI. When UA1 invited UA2 via openser, openser only ask A record of UA2. There are no A records in my DNS server! So, openser then replied 476 Unresolvable Destination message to UA1. In normal conditon, if there were no such name of A record, it should ask AAAA record instead. But why OPENSER only ask A record? Best Regards, Fang

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Re: [Serusers] PRACK on forked response
by Stephen Paterson 15 Jul '08

15 Jul '08

Don't worry about this, I've found the problem and it was nothing to do with SER - the reliable 1xx was being sent without a contact header so the PRACK was being sent with the same RURI as the initial INVITE. Nothing SER could do but fork the PRACK. All working now I've fixed the UAS. Cheers Steve -----Original Message----- From: Stephen Paterson Sent: 15 July 2008 10:21 To: 'Michal Matyska' Cc: serusers(a)iptel.org Subject: RE: [Serusers] PRACK on forked response Hi Michal, Thanks for the reply and sorry for the delay. Had me a nice relaxing long weekend in the country! Anyhow, I've not set up anything special in the config file. I just installed SER and ran it straight out of the box as it were. It worked fine for my purposes so I felt no need to tinker with the config. There is no mention of forking or lookup_contacts in the config. Diff shows the only difference between my config file and the simple example one that comes with the distribution is at the top in the global configuration parameters section where I set a listen address. All the routing logic is identical. Cheers Steve -----Original Message----- From: Michal Matyska [mailto:michal.matyska@iptel.org] Sent: 09 July 2008 15:11 To: Stephen Paterson Cc: serusers(a)iptel.org Subject: Re: [Serusers] PRACK on forked response Hi, SER does not care about PRACK... it is just another within-dialog request type, like BYE and many others. I think you have incorrect routing script logic, where you fork even the within dialog requests (using lookup_contacts) instead of just following the record route set. Michal Stephen Paterson píše v St 09. 07. 2008 v 14:01 +0100: > Hi all, > > I'm using SER purely for testing purposes just now. Currently testing > our handling of forked responses and hit upon the following situation: > > My outgoing INVITE is forked to two UASs (also our own SIP endpoint), > one of which sends a reliable 180 and the other an unreliable 180. The > UAC sends a PRACK on the dialog ID of the reliable response but SER > forwards this response on to both UASs the result being the UAS that > sent the unreliable 180 responds to the PRACK with a 481 (the PRACK > contains a To-tag that part defines a dialog unknown to that UAS). I > confess I can't find anything describing this scenario in any RFC but > it strikes me as incorrect behaviour on SERs part. > > Does SER support PRACK? If not then no matter, I can live with it - it > doesn't affect the outcome of any call. If it does, does anyone have > any thoughts on this? > > Cheers > > Steve > > Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, > MK1 1PT, UK > Registration No: 1397386 (Wales) > > P Please consider the environment and don't print this e-mail unless > you really need to > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales) P Please consider the environment and don't print this e-mail unless you really need to

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[OpenSER-Users] SIMPLE-XMPP Developer Workshop 2008
by Daniel-Constantin Mierla 15 Jul '08

15 Jul '08

Hello, for those interested in SIP/SIMPLE - XMPP(Jabber/GTalk) interoperability, there is going to be a workshop in Paris for developers in the first week of September. Agenda for openser is to identify a reliable xmpp library that follows latest XMPP specs and offers secure communication (TLS) which can be used to update and enhance the simple-xmpp gateway implementation. In addition, it is a good opportunity to deploy and test it against other implementations. If you have extra ideas for openser sip-xmpp, please write me or, better, join the event and help get it done. Detailed description of the event follows. Best regards, Daniel SIMPLE-XMPP Developer Workshop 2008 SIP/SIMPLE and XMPP share a lot of concepts but they are different in many aspects. Being nowadays the leading open protocols for voice, video, instant messaging and presence, the interconnection between them creates an unified communication environment for users in both sides. The workshop aims to bring together people with large expertise in both protocols, interested in development, testing and deployment of SIP/SIMPLE-XMPP solutions. With a permanent focus on innovation, the participants cover open source projects to private enterprises. You can join the event for free. Date: September 2-5, 2008 Place: INRIA, Paris, France Main organizers: - Philippe Sultan, INRIA - contributor to the XMPP/Jingle support in Asterisk, member of XMPP Standards Foundation - Olle E. Johansson, Edvina - main SIP developer of Asterisk - Daniel-Constantin Mierla, Asipto - co-founder Openser, developer of XMPP/Jabber gateway Goal: - kick up simple-xmpp interoperability Agenda guidelines: - identify common issues of simple-xmpp interoperability - define best-practice solutions and workarounds of delicate issues - coding sessions in existing applications such as Asterisk, Openser, Jabberd, Freeswitch, eJabberd, libraries, client applications, etc. - testing sessions - reports about past experiences and results of the workshop Target participants: - developers of simple-xmpp products - people interested in testing simple-xmpp products - people interested in building simple-xmpp communication environments Cost: - free registration (everybody pays for its traveling and accommodation) Registration: - via e-mail at simple-xmpp(a)asipto.com - please write the motivation to participate and add bullets into agenda if you like to approach new subjects. The organizers reserve the right to select a group of people that will contribute most as this is mainly a workshop to approach issues, test interoperability and design new solutions. - updates about the event are posted at: http://www.asipto.com/index.php/simple-xmpp-developer-workshop-2008/ Size: - up to 20 people As of Jul 15, 2008, participating companies are: - INRIA - http://www.inria.fr - SIP-XMPP interoperability in Asterisk (http://www.asterisk.org) - Edvina - http://www.edvina.net - SIP-XMPP interoperability in Asterisk (http://www.asterisk.org) - AG Projects - http://www.ag-projects.com - MSRP-XMPP Interoperability (http://www.msrprelay.org) - Asipto - http://www.asipto.com - SIP-XMPP interoperability in Openser (http://www.openser.org) General discussions about sip-xmpp can be conducted via mailing list: sip-xmpp [at] xmpp.org http://mail.jabber.org/mailman/listinfo/sip-xmpp -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] Limit call duration
by inge 15 Jul '08

15 Jul '08

Hi all, I would like to limit the call duration for two hours. Ideally, limitation could be drive by Seconds or Euros. Indeed, for example 10€ for mobile destination (0.1€/min for example) can allow the user to call during 100min. I have thinking of using SIPSAK to generate a BYE but I need something between SER and SIPSAK to do the calculation. What could be a workarround to do that ? Regards, Adrien .L

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[Serusers] PRACK on forked response
by Stephen Paterson 15 Jul '08

15 Jul '08

Hi all, I'm using SER purely for testing purposes just now. Currently testing our handling of forked responses and hit upon the following situation: My outgoing INVITE is forked to two UASs (also our own SIP endpoint), one of which sends a reliable 180 and the other an unreliable 180. The UAC sends a PRACK on the dialog ID of the reliable response but SER forwards this response on to both UASs the result being the UAS that sent the unreliable 180 responds to the PRACK with a 481 (the PRACK contains a To-tag that part defines a dialog unknown to that UAS). I confess I can't find anything describing this scenario in any RFC but it strikes me as incorrect behaviour on SERs part. Does SER support PRACK? If not then no matter, I can live with it - it doesn't affect the outcome of any call. If it does, does anyone have any thoughts on this? Cheers Steve Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales) P Please consider the environment and don't print this e-mail unless you really need to

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[OpenSER-Users] Registration redirect
by David Villasmil 15 Jul '08

15 Jul '08

Hello all! Lets say I have domain sip.xyz.com directed to IP 1.2.3.4. But I want customers to register and call usgin ip 9.8.7.6 which is NOT in the dns list for sip.xyz.com. If i'm not mistaken, doing a simple redirect I should be able to get my customers registered on 9.8.7.6. But is there any other considerations I should make? thanks! d

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[OpenSER-Users] Adam Kusmirek : nie ma mnie w pracy.
by Adam Kusmirek 15 Jul '08

15 Jul '08

Nie będzie mnie w pracy od 2008-07-14 i nie wrócę przed 2008-07-28. Podczas nieobecności zastępuje mnie Tomek Mazan.

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[OpenSER-Users] For Openser To Background
by Ali Jawad 15 Jul '08

15 Jul '08

Hi All When I start Openser it starts in the putty console and the output is displayed there, my softphone can logon during that time. If I do close that putty session my softphone can not logon anymore although if I do netstat it states that Openser is listening. Howe can I send openser to the background? Or make it work in daemon mode ? My config has this info debug=9 log_stderror=yes log_facility=LOG_LOCAL0 #fork=yes fork=no children=4 Thanks

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[Serusers] Hello, request of the module for RFC 4474 Identity authentication
by Ge Zhang 15 Jul '08

15 Jul '08

Hello, everyone, I would like to install SER2.0 with the module of RFC 4474 Identity authentication. Could you please tell me which module implemented this if you know? Thanks a lot! Best regards, Ge

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[OpenSER-Users] Openser - which b2bua to use ?
by Amit Jain 15 Jul '08

15 Jul '08

I have configured Openser and integrated with radius server. I have done auth and accounting too. Now i need some help for b2bua to terminate the session after prepaid credit is over. which is the recommended b2bua that i can use for my configuration. Please let me know, how to proceed further. Regards, Amit

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[OpenSER-Users] Contacts in 3xx message on one line or multiple lines?
by Hiers, David 14 Jul '08

14 Jul '08

Hi, Openser seems to be happy with multiple contacts on one line in a 302 message. Contact:<sip:5551234567@255.16.250.13:5060;user=phone>;q=0.5,<sip:555123 4567@255.16.250.10:5060;user=phone>;q=0.25 If the leftmost contact is down, the rightmost contact is tried as expected. However, Openser does not like multiple lines of one contact per line: Contact: <sip:5551234567@255.16.250.16:5060;dtg=SIP-NS> Contact: <sip:5551234567@255.16.250.19:5060;dtg=SIP-NS> Openser uses the topmost contact and ignores the bottommost contact in the multiple line list. If the topmost contact is down, the bottommost is never tried. Is Openser supposed to be able to handle multiline contact lists? If so, what are the common configuration issues that we can expect to see? Thanks, David Hiers CCIE (R/S, V), CISSP ADP Dealer Services 2525 SW 1st Ave. Suite 300W Portland, OR 97201 o: 503-205-4467 f: 503-402-3277 This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

2 1

[Serusers] checking permissions with if (!$f.gw_acl=="1")
by Davor Jovanovic 14 Jul '08

14 Jul '08

Hi all, I'm using SER2.0 (testing) and I have a question regarding checking the acl AVP to verify that the user is allowed to call PSTN destinations with if (!$f.gw_acl=="1") { ... In the "user_attrs" table I have this entry: uid name value type flags alice(a)xyz.hr gw_acl 1 2 61 before checking this "gw_acl" I do load attributes with load_attrs("$fu", "$fu.uid"); but I keep entering above "if" statement. So I would appreciate any help about how to solve this issue, in which tables I need to enter which values... I have solved this checking with avpops module and "avp_check" function, but as I already use new AVP schema in SER2.0 for loading other user attributes I would like to use it with this "gw_acl" also. Thanks /davor

2 2

[OpenSER-Users] Problems with BYE
by Jan P 14 Jul '08

14 Jul '08

Hi, I made a pretty basic openser config with Sipwise but Im having problems when sending BYE to my upstream SIP provider. When I do a CANCEL it works but sending BYE the server doesnt forward any packets. Here's a ngrep of whats happening. My complete openser.cfg can be found at http://port53.se/openser.txt # U 99.88.164.203:30824 -> 101.12.20.26:5060 BYE sip:0709955324@bbr-ssw.sto.la.aa.net SIP/2.0. Via: SIP/2.0/UDP 10.45.0.98:30824;branch=z9hG4bK-d87543-f13c745a11112e7c-1--d87543-;rport. Max-Forwards: 70. Route: <sip:101.12.20.26;lr;ftag=9a47eb12;nat=yes>. Route: <sip:sipgw.sds.se;lr>. Contact: <sip:0722465930@99.88.164.203:30824>. To: "0709955324"<sip:0709955324@proxy.realm.se>;tag=ecbc244f. From: "0722465930"<sip:0722465930@proxy.realm.se>;tag=9a47eb12. Call-ID: ZDYwNjNjMmYzMTdiZGE3NTc4Nzc4OTA2NjdkOTllYTU.. CSeq: 3 BYE. Proxy-Authorization: Digest username="30000",realm="proxy.realm.se",nonce="487473f9db3e3da3ae9c3115a6d87d2023cc6b0a",uri="sip:0709955324@bbr-ssw.sto.la.aa.net",response="d9dc90e88deba0d9ba428bc11554b193",algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Reason: SIP;description="User Hung Up". Content-Length: 0. . # U 101.12.20.26:5060 -> 99.88.164.203:30824 SIP/2.0 500 I'm terribly sorry, server error occurred (7/TM). Via: SIP/2.0/UDP 10.45.0.98:30824;branch=z9hG4bK-d87543-f13c745a11112e7c-1--d87543-;rport=30824;received=99.88.164.203. To: "0709955324"<sip:0709955324@proxy3.digisip.net>;tag=ecbc244f. From: "0722465930"<sip:0722465930@proxy3.digisip.net>;tag=9a47eb12. Call-ID: ZDYwNjNjMmYzMTdiZGE3NTc4Nzc4OTA2NjdkOTllYTU.. CSeq: 3 BYE. Server: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. . #

2 3

Re: [OpenSER-Users] FW: OpenSER 1.3.2-tls - error tcp_blocking_connect
by Bogdan-Andrei Iancu 14 Jul '08

14 Jul '08

TLS uses a different port than TCP. TCP is 5060 and TLS is 5061. Regards, Bogdan pravin.swami(a)wipro.com wrote: > Hi Bogdan-Andrei Iancu, > > Thanks for your reply . > > When Sending a INVITE request without (no-TLS) proxy establishes TCP > connection successfully & call Succeeds. > But, when tried INVITE with (TLS) proxy fails to establish TCP > connection & get the below error. > > Am I missing something?? > Is there any configurable parameter which has to be used in ser.cfg > > Regards, > Pravin.swami > > > > > > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, July 14, 2008 2:02 PM > To: Pravin Sidram swami (WT01 - MCE-Mobile Devices) > Cc: users(a)lists.openser.org > Subject: Re: [OpenSER-Users] FW: OpenSER 1.3.2-tls - error > tcp_blocking_connect > > Hi, > > The error message specks for it self : > Connection refused > > So, your OpenSER tried to open a TCP/TLS connection to an address where > nobody was listening. > > Regards, > Bogdan > > > pravin.swami(a)wipro.com wrote: > >> >> Hi... All , >> >> I have Configured OpenSER 1.3.2-tls source code & compiled >> successfully . >> >> Registration is successful with TLS handshake . >> >> Below is the scenario of my test . >> >> UA1 & UA2 clients Registers on TLS. >> >> When UA1 send INVITE request to UA2 >> >> I got the Following Error : >> >> : SIP/2.0 477 Send failed (477/TM) >> >> When i scanned through the logs of openser I got the following Message >> > > >> : proxy always trying to establish TCP connection but it Fails. >> >> Jul 11 23:55:20 [22415] DBG:core:check_via_address: params >> 10.200.68.220, 10.200.68.220, 0 Jul 11 23:55:20 [22415] >> DBG:core:tcp_send: no open tcp connection found, opening new one Jul >> 11 23:55:20 [22415] ERROR:core:tcp_blocking_connect: poll error: >> flags 18 >> Jul 11 23:55:20 [22415] ERROR:core:tcp_blocking_connect: failed to >> retrieve SO_ERROR (111) Connection refused Jul 11 23:55:20 [22415] >> ERROR:core:tcpconn_connect: >> tcp_blocking_connect failed >> Jul 11 23:55:20 [22415] ERROR:core:tcp_send: connect failed Jul 11 >> 23:55:20 [22415] ERROR:tm:msg_send: tcp_send failed Jul 11 23:55:20 >> [22415] ERROR:tm:t_forward_nonack: sending request failed Jul 11 >> 23:55:20 [22415] DBG:tm:t_relay_to: t_forward_nonack returned error >> Jul 11 23:55:20 [22415] DBG:core:parse_headers: flags=ffffffffffffffff >> > > >> Jul 11 23:55:20 [22415] DBG:core:check_via_address: params >> 10.200.68.220, 10.200.68.220, 0 Jul 11 23:55:20 [22415] >> DBG:tm:run_trans_callbacks: trans=0xb5db7e50, callback type 128, id 0 >> entered >> ACC: call missed: >> timestamp=1215800720;method=INVITE;from_tag=20282693445eido3yklvo4;to_ >> tag=;call_id=8307080595eido3erlzgw;code=477;reason=Request >> Failure >> >> - Presently i don't have any DNS server , is that >> really matters ?? >> - While openser tries to open TCP connection with >> UAS it's sending port ' 0 ' , but fails is it ok .?? , or Anything >> I'm missing in Configuration . >> I have Attahced config file Also. >> >> Thanks & Regards, >> Pravin.swami >> >>

1 0

[OpenSER-Users] Need Help on RLS.
by mahesh.peddi＠info-spectrum.com 14 Jul '08

14 Jul '08

Hi Experts, I need information on RLS. I have downloaded some material on RLS from internet. I have read document ,Though i got brief idea on this.I am not clear on how to proceed. I have to develop the code from scratch or and RPM is available. please suggest me on this. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

2 1

[OpenSER-Users] asterisk/openser/mysql
by Eliass Salim 14 Jul '08

14 Jul '08

Hi, I've installed asterisk on a computer and openser on an other, I already have a mysql database for openser, and I'm now trying to install one for asterisk, but when I type: CREATE DATABASE asterisk; USE asterisk; CREATE VIEW voicemail AS SELECT phplib_id as uniqueid, username as customer_id, 'default' as context, username as mailbox, vmail_password as password, CONCAT(first_name,' ',last_name) as fullname, email_address as email, NULL as pager, datetime_created as stamp FROM openser.subscriber WHERE vmail = TRUE; I have the following error: ERROR 1054 (42S22): Unknown column 'phplib_id' in 'field list' I already had this problem for opensertcl, but I fixed it by adding WEBSER="yes" on openserctlr. thanks, Eliass

3 8

[OpenSER-Users] FW: OpenSER 1.3.2-tls - error tcp_blocking_connect
by pravin.swami＠wipro.com 14 Jul '08

14 Jul '08

Hi... All , I have Configured OpenSER 1.3.2-tls source code & compiled successfully . Registration is successful with TLS handshake . Below is the scenario of my test . UA1 & UA2 clients Registers on TLS. When UA1 send INVITE request to UA2 I got the Following Error : : SIP/2.0 477 Send failed (477/TM) When i scanned through the logs of openser I got the following Message : proxy always trying to establish TCP connection but it Fails. Jul 11 23:55:20 [22415] DBG:core:check_via_address: params 10.200.68.220, 10.200.68.220, 0 Jul 11 23:55:20 [22415] DBG:core:tcp_send: no open tcp connection found, opening new one Jul 11 23:55:20 [22415] ERROR:core:tcp_blocking_connect: poll error: flags 18 Jul 11 23:55:20 [22415] ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Jul 11 23:55:20 [22415] ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jul 11 23:55:20 [22415] ERROR:core:tcp_send: connect failed Jul 11 23:55:20 [22415] ERROR:tm:msg_send: tcp_send failed Jul 11 23:55:20 [22415] ERROR:tm:t_forward_nonack: sending request failed Jul 11 23:55:20 [22415] DBG:tm:t_relay_to: t_forward_nonack returned error Jul 11 23:55:20 [22415] DBG:core:parse_headers: flags=ffffffffffffffff Jul 11 23:55:20 [22415] DBG:core:check_via_address: params 10.200.68.220, 10.200.68.220, 0 Jul 11 23:55:20 [22415] DBG:tm:run_trans_callbacks: trans=0xb5db7e50, callback type 128, id 0 entered ACC: call missed: timestamp=1215800720;method=INVITE;from_tag=20282693445eido3yklvo4;to_ta g=;call_id=8307080595eido3erlzgw;code=477;reason=Request Failure - Presently i don't have any DNS server , is that really matters ?? - While openser tries to open TCP connection with UAS it's sending port ' 0 ' , but fails is it ok .?? , or Anything I'm missing in Configuration . I have Attahced config file Also. Thanks & Regards, Pravin.swami Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[OpenSER-Users] ERROR:registrar:update_contacts: invalid cseq for aor
by Dilip 14 Jul '08

14 Jul '08

Hello evryone, I am using openser-1.1.1-notls version. I am using auth_radius module for the registratrion. When i was firing the call continues then after some time openser gives me the error like ERROR:registrar:update_contacts: invalid cseq for aor <8885073>. This error comes continuesly and bcz of that retramision is increasing. I have used only this paramater of userloc module modparam("usrloc", "db_mode", 2) So what may be problem?? Can anybody help me to solve the problem. Can u explain me. Regards, Dilip

2 1

[OpenSER-Users] T.38 and RTP Read too short
by Jens Carl 14 Jul '08

14 Jul '08

Hey everybody, I use OpenSER (openser 1.3.2-tls (x86_64/linux), svnrevision: 2:4420) and the RTPproxy (1.1) and everything works fine expect for one thing. If I try to use T.38 with following scenario: SIP-FAX1 (NAT) <-> RTPProxy+OpenSER <-> Asterisk <-> Carrier the Asterisk always produce this kind of warning: 'WARNING[1493]: rtp.c:1145 ast_rtp_read: RTP Read too short'. The SIP-signaling with the re-INVITE works fine. If I get an re-INVITE I call the force_rpt_proxy( 'l' ). Has anybody an idea what I should change in the RTPproxy to get this scenario working? Thanks for your help, Jens

3 4

[Serusers] RTP-proxy behind NAT
by vicos hack 13 Jul '08

13 Jul '08

hello, I was wondering if any one have had any experience on making RTP-proxy work behind a Firewall-NAT-Router like in this: [Nated Clients] | | [ROUTER--NAT--FIREWALL] (192.168.5.1 / 77.95.143.30) | [ Openser/RTP-Proxy] (192.168.5.40) uses Dispatcher module to distribute calls to asterisk servers | | | | [ASTERISK]1(.41) [ASTERISK2](.42) | | | | --------[T1/PRI/SIP/H323]----- if so ,,how can this be achieve ? Thanks Vicox

1 0

[Serusers] OpenSER 1.3.2-tls - error tcp_blocking_connect
by pravin.swami＠wipro.com 13 Jul '08

13 Jul '08

Hi... All , I have Configured OpenSER 1.3.2-tls source code & compiled successfully . Registration is successful with TLS handshake . Below is the scenario of my test . UA1 & UA2 clients Registers on TLS. When UA1 send INVITE request to UA2 I got the Following Error : : SIP/2.0 477 Send failed (477/TM) When i scanned through the logs of openser I got the following Message : proxy always trying to establish TCP connection but it Fails. Jul 11 23:55:20 [22415] DBG:core:check_via_address: params 10.200.68.220, 10.200.68.220, 0 Jul 11 23:55:20 [22415] DBG:core:tcp_send: no open tcp connection found, opening new one Jul 11 23:55:20 [22415] ERROR:core:tcp_blocking_connect: poll error: flags 18 Jul 11 23:55:20 [22415] ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Jul 11 23:55:20 [22415] ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jul 11 23:55:20 [22415] ERROR:core:tcp_send: connect failed Jul 11 23:55:20 [22415] ERROR:tm:msg_send: tcp_send failed Jul 11 23:55:20 [22415] ERROR:tm:t_forward_nonack: sending request failed Jul 11 23:55:20 [22415] DBG:tm:t_relay_to: t_forward_nonack returned error Jul 11 23:55:20 [22415] DBG:core:parse_headers: flags=ffffffffffffffff Jul 11 23:55:20 [22415] DBG:core:check_via_address: params 10.200.68.220, 10.200.68.220, 0 Jul 11 23:55:20 [22415] DBG:tm:run_trans_callbacks: trans=0xb5db7e50, callback type 128, id 0 entered ACC: call missed: timestamp=1215800720;method=INVITE;from_tag=20282693445eido3yklvo4;to_ta g=;call_id=8307080595eido3erlzgw;code=477;reason=Request Failure - Presently i don't have any DNS server , is that really matters ?? - While openser tries to open TCP connection with UAS it's sending port ' 0 ' , but fails is it ok .?? , or Anything I'm missing in Configuration . I have Attahced config file Also. Thanks & Regards, Pravin.swami Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Serusers] Modified Document about Solaris10_plus_SER-0.9.6_plus_SERweb-0.9.6
by muhammad akl 13 Jul '08

13 Jul '08

Hey All, well, more than one week i sent an e-mail to serusers mailing list about iptel's documentation outdated and asked peoples that who can help me in writing some topics related specifically to solaris10 os , but didn't get any response about that. so as usual do it myself and since i have rewitten this tutorial from ground i will update it frequently this new updated version include : 1- installing coolstack package for solaris and configuring it 2- installing SER version 0.9.6 3-installing SERweb version 0.9.6 thins TBD : 1-installing more configuration files like working with nat, pstn 2-installing SEMS and configure it to work with SER 3-integrate SER to work together with Asterisk for sems with ser i need really who can help me to do that and for the rest points i'll try my best to make them to real heres the document : http://muhammad.akl.googlepages.com/Solaris10-SER.0.9.6-SERweb-0.9.6.pdf the document is still in beta version so suggestions, corrections and feedbacks are welcomed. BR muhammad

4 5

Re: [Serusers] usrloc won't work after a couple days?
by lin9k＠yahoo.com 13 Jul '08

13 Jul '08

Hi Martin, Thanks for replying. I am using ser-0.9.6. How do I check the period of silence? I have also noticed that alias cannot be recorded/updated into mysql too. Looks like the problem is as you were referring. Which version has the fix? Where is the culprit? Ling ----- Original Message ---- From: Martin Hoffmann <hn(a)nvnc.de> To: lin9k(a)yahoo.com Cc: serusers(a)lists.iptel.org Sent: Tuesday, July 1, 2008 11:33:58 PM Subject: Re: [Serusers] usrloc won't work after a couple days? lin9k(a)yahoo.com wrote: > > I have setup ser to run with mysql. At beginning, it worked fine that > the mysql/ser/location table got the correct update. However, after 3 > days the table won't get updated. When I enable the mysql log I cannot > see any updating or inserting into that table. User registration is > still working fine but just not get updated into the location table. > Also, server_monitoring tables are working fine. Do you have a longer period of silence in between? I seem to remember that there was a problem with the connection between SER and MySQL getting shut down and SER not reconnecting. Dunno if this was in the SER version you are using, though. Regards, Martin

3 2

[Serusers] Solaris10_plus_SER-0.9.6_plus_SERweb-0.9.6
by Do Nguyen Ha 11 Jul '08

11 Jul '08

Hi Muhammad, > > On Thu, Jul 10, 2008 at 5:17 PM, muhammad akl <muhammad.akl(a)gmail.com> > wrote: >> This new updated version include : >> >> 1- installing coolstack package for solaris and configuring it >> 2- installing SER version 0.9.6 >> 3-installing SERweb version 0.9.6 thanks for great help the docs for SER is out date, that is why alot of people need help SER is stable but docs is not good for someone starting Openser is good docs for it Thanks

2 1

[OpenSER-Users] ipv4 and ipv6 intercommunication
by 苗杰 10 Jul '08

10 Jul '08

I'm a beginner at openser. What does the variable 'af' stand for?And what is the value of the optional? Thanks! 313 # Comment out three lines below if you want 314 # RTP for IPv4->IPv4 calls to go directly 315 # between UAs 316 if (af == inet) { 317 force_rtp_proxy("faii"); 318 t_on_reply("1"); 319 } 320 # proxy session from a Internal IPv4 321 # phone to a External IPv6 address 322 if (af == inet6) { 323 force_rtp_proxy("faie"); 324 t_on_reply("1"); 325 } 326 } 327 328 else if (lookup("location_inet6")) { 329 # proxy session from a External IPv6 330 # phone to a Internal IPv4 address 331 if (af == inet) { 332 force_rtp_proxy("faei"); 333 t_on_reply("1"); 334 } 335 336 # Comment out three lines below if you want 337 # RTP for IPv6->IPv6 calls to go directly 338 # between UAs 339 if (af == inet6) { 340 force_rtp_proxy("faee"); 341 t_on_reply("1"); 342 }

2 1

[OpenSER-Users] IPv6 and IPv4
by 苗杰 10 Jul '08

10 Jul '08

313 # Comment out three lines below if you want 314 # RTP for IPv4->IPv4 calls to go directly 315 # between UAs 316 if (af == inet) { 317 force_rtp_proxy("faii"); 318 t_on_reply("1"); 319 } 320 # proxy session from a Internal IPv4 321 # phone to a External IPv6 address 322 if (af == inet6) { 323 force_rtp_proxy("faie"); 324 t_on_reply("1"); 325 } 326 } 327 328 else if (lookup("location_inet6")) { 329 # proxy session from a External IPv6 330 # phone to a Internal IPv4 address 331 if (af == inet) { 332 force_rtp_proxy("faei"); 333 t_on_reply("1"); 334 } 335 336 # Comment out three lines below if you want 337 # RTP for IPv6->IPv6 calls to go directly 338 # between UAs 339 if (af == inet6) { 340 force_rtp_proxy("faee"); 341 t_on_reply("1"); 342 }

3 2

[OpenSER-Users] fix_nated_register() Error.
by Ali Jawad 10 Jul '08

10 Jul '08

Hi All I am getting this error on Openser 1.3 Jul 10 11:28:31 [20126] ERROR:nathelper:fixup_fix_nated_register: you must set 'received_avp' parameter. Must be same value as parameter 'received_avp' of registrar module Jul 10 11:28:31 [20126] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 386 Jul 10 11:28:31 [20126] ERROR:core:main: failed to fix configuration with err code -1 I used the example from nat-traversal.cfg I do believe the error is initiated from this section route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } Please advice.

3 6

[OpenSER-Users] ipv6 parsing error
by Andhika Gannesha Gemilang 09 Jul '08

09 Jul '08

Hi, Im trying to setup ipv6 for my openser. i add this line in my openser.cfg *LIsten=udp:[2004:fa22:337:4::81]:5060 Listen=tcp:[2004:fa22:337:4::81]:5060 *and then when im trying to register with my sip phone, i got these line: ERROR:core:parse_uri: bad char ':' in state 3 parsed: <sip:2004:fa22> (13) / <sip:2004:fa22:337:4::81> (23) Jul 8 12:02:52 [2977] ERROR:core:parse_sip_msg_uri: bad uri <sip:2004:fa22:337:4::81> Jul 8 12:02:52 [2977] WARNING:core:do_action: error in expression Jul 8 12:02:52 [2981] ERROR:core:parse_uri: bad char ':' in state 3 parsed: <sip:2004:fa22> (13) / <sip:2004:fa22:337:4::81> (23) Jul 8 12:02:52 [2981] ERROR:registrar:extract_aor: failed to parse Address of Record Jul 8 12:02:52 [2981] ERROR:registrar:save: failed to extract Address Of Record and these from tcpdump: 12:11:52.267529 IP6 2004:fa22:337:4::84.sip > 2004:fa22:337:4::81.sip: SIP, length: 488 12:11:52.272578 IP6 2004:fa22:337:4::81 > ff02::1:ff00:84: ICMP6, neighbor solicitation, who has 2004:fa22:337:4::84, length 32 12:11:52.273059 IP6 2004:fa22:337:4::84 > 2004:fa22:337:4::81: ICMP6, neighbor advertisement, tgt is 2004:fa22:337:4::84, length 32 12:11:52.273068 IP6 2004:fa22:337:4::81.sip > 2004:fa22:337:4::84.sip: SIP, length: 500 12:11:52.469341 IP6 2004:fa22:337:4::84.sip > 2004:fa22:337:4::81.sip: SIP, length: 671 12:11:52.470127 IP6 2004:fa22:337:4::81.sip > 2004:fa22:337:4::84.sip: SIP, length: 445 12:11:57.264948 IP6 fe80::213:20ff:feac:ce4f > 2004:fa22:337:4::81: ICMP6, neighbor solicitation, who has 2004:fa22:337:4::81, length 32 also, i have disabled the ip6tables. any assistance would be so helpful to me thankyou,

4 7

[OpenSER-Users] Is there a REFER function on Openser
by mark morreny 09 Jul '08

09 Jul '08

Hi, I am looking for a REFER function which can 1) redirect UA to another sip server and 2) get openser out of the hoop. Does Openser has this kind of feature? Thanks for all your help. Mark

2 1

[OpenSER-Users] ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo
by suresh tattikota 09 Jul '08

09 Jul '08

hi all i am trying to add a module pua to Openser using loadmodule "pua.so" in openser.cfg i am geting some errors like this [root@info ~]# *openserctl moni* database engine 'MYSQL' loaded Control engine 'FIFO' loaded ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have line 'fifo=/tmp/openser_fifo' in your config please help me regarding this thank u

2 1

[OpenSER-Users] About Openser Message
by Ali Jawad 09 Jul '08

09 Jul '08

Hi All hi when I do a call on Openser it shows the following on my softphone CallStatus Message: 183 SESSION PROGRESS;Ringing-tone=TRUE; Can this message be changed, if so how? Thanks

1 0

[OpenSER-Users] INVITE NAT problem
by Juan Backson 09 Jul '08

09 Jul '08

Hi, I am working on getting my two SIP phones behind firewall to talk to each other. One client is SPA 921 and the other client is xlite; and I am force rport for all SIP requests from clients. When xlite sends a INVITE request to openser, openser tries to send to SPA 921, but the PORT is still 5060, instead of 22848. Therefore, Openser keeps resending INVITE to SPA 921 but never reaches it. Both linksys and xlite do send preiodic SUBSCRIBE/NOTIFY to Openser. Is there anything else I am missing? U 48.133.23.45:21395 -> 48.133.200.20:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:53722;branch=z9hG4bK-d87543-683f775fdd79a93d-1--d 87543-;rport. Max-Forwards: 70. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O. Content-Type: application/sdp. Supported: timer. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 371. . v=0. o=- 9 2 IN IP4 192.168.1.100. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.100. t=0 0. m=audio 5416 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. U 48.133.200.20:5060 -> 48.133.23.45:21395 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.1.100:53722;branch=z9hG4bK-d87543-683f775fdd79a93d-1--d 87543-;rport=21395;received=48.133.23.45. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 48.133.200.20:5060 -> 48.133.23.45:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Record-Route: <sip:48.133.200.20;lr=on;ftag=12709c02>. Via: SIP/2.0/UDP 48.133.200.20;branch=z9hG4bK0996.d020a552.0. Via: SIP/2.0/UDP 192.168.1.100:53722;received=48.133.23.45;branch=z9hG4bK-d8754 3-683f775fdd79a93d-1--d87543-;rport=21395. Max-Forwards: 69. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, U 48.133.23.45:5060 -> 48.133.200.20:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Record-Route: <sip:48.133.200.20;lr=on;ftag=12709c02>. Via: SIP/2.0/UDP 48.133.200.20;branch=z9hG4bK0996.d020a552.0. Via: SIP/2.0/UDP 192.168.1.100:53722;received=48.133.23.45;branch=z9hG4bK-d8754 3-683f775fdd79a93d-1--d87543-;rport=21395. Max-Forwards: 69. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, U 48.133.200.20:5060 -> 48.133.23.45:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Record-Route: <sip:48.133.200.20;lr=on;ftag=12709c02>. Via: SIP/2.0/UDP 48.133.200.20;branch=z9hG4bK0996.d020a552.0. Via: SIP/2.0/UDP 192.168.1.100:53722;received=48.133.23.45;branch=z9hG4bK-d8754 3-683f775fdd79a93d-1--d87543-;rport=21395. Max-Forwards: 69. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, U 48.133.23.45:5060 -> 48.133.200.20:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Record-Route: <sip:48.133.200.20;lr=on;ftag=12709c02>. Via: SIP/2.0/UDP 48.133.200.20;branch=z9hG4bK0996.d020a552.0. Via: SIP/2.0/UDP 192.168.1.100:53722;received=48.133.23.45;branch=z9hG4bK-d8754 3-683f775fdd79a93d-1--d87543-;rport=21395. Max-Forwards: 69. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, U 48.133.200.20:5060 -> 48.133.23.45:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Record-Route: <sip:48.133.200.20;lr=on;ftag=12709c02>. Via: SIP/2.0/UDP 48.133.200.20;branch=z9hG4bK0996.d020a552.0. Via: SIP/2.0/UDP 192.168.1.100:53722;received=48.133.23.45;branch=z9hG4bK-d8754 3-683f775fdd79a93d-1--d87543-;rport=21395. Max-Forwards: 69. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, U 48.133.23.45:5060 -> 48.133.200.20:5060 INVITE sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> SIP/2.0. Record-Route: <sip:48.133.200.20;lr=on;ftag=12709c02>. Via: SIP/2.0/UDP 48.133.200.20;branch=z9hG4bK0996.d020a552.0. Via: SIP/2.0/UDP 192.168.1.100:53722;received=48.133.23.45;branch=z9hG4bK-d8754 3-683f775fdd79a93d-1--d87543-;rport=21395. Max-Forwards: 69. Contact: <sip:2000@48.133.23.45:21395>. To: "1005"<sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=12709c02. Call-ID: MzRjM2RkZjQxNDk3ZjVjOTFhMzYyZmM0MThmODI3N2U.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, U 48.133.23.45:21395 -> 48.133.200.20:5060 SUBSCRIBE sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:53722;branch=z9hG4bK-d87543-d900ee1c55767008-1--d 87543-;rport. Max-Forwards: 70. Contact: <sip:2000@48.133.23.45:21395>. To: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>. From: "2000"<sip:2000@48.133.23.45 <sip%3A2000(a)48.133.23.45>>;tag=8b347a51. Call-ID: MzY3NGIwNTJlMWY5Njg2MmM5YmI1NTIxNTRjZjljYzM.. CSeq: 1 SUBSCRIBE. Expires: 300. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O. Supported: timer. User-Agent: X-Lite release 1011s stamp 41150. Event: message-summary. Content-Length: 0. . U 48.133.23.45:22840 -> 48.133.200.20:5060 NOTIFY sip:48.133.23.45 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK-2c2e52f6;rport. From: "1005" <sip:1005@48.133.23.45 <sip%3A1005(a)48.133.23.45> >;tag=a45646112863dco0. To: <sip:48.133.23.45>. Call-ID: ad4c4ce-bb2b86f4(a)192.168.1.101. CSeq: 1017 NOTIFY. Max-Forwards: 70. Event: keep-alive. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. . Any suggestion will be helpful. JB

3 2

[OpenSER-Users] [MediaProxy] Sessions ends ("did timeout") during the INVITE
by Iñaki Baz Castillo 09 Jul '08

09 Jul '08

Hi, I'm suffering an important issue with MediaProxy in calls from/to my carrier softswitch. Sometimes it seems that MediaProxy assigns a media port during the INVITE but it closes it inmediatelly, so the following 183 forces the creation of a different media port and obviously the media dies. It works perfectly for long days but some days it has those problems. Please, could you people help me in investigating if a problem in my OpenSer/MediaProxy? I have a detailed real example. This is what happens: --------------------------------------------------------------------------- 09:28:46 - INVITE Client -> OpenSer (audio: 192.168.1.203:10370) 09:28:46 - INVITE OpenSer -> gateway (audio: 88.99.3.10:60696) <-- MediaProxy set 60696 port but inmediatelly ends the session. 09:28:48 - 183 Gateway -> OpenSer (audio: 66.11.6.142:45734) 09:28:48 - 183 OpenSer -> Client (audio: 88.99.3.10:60698) <-- Wrong port, MediaProxy set 60698 port !!! 09:28:49 - 200 Gateway -> OpenSer (audio: 66.11.6.142:45734) 09:28:49 - 200 OpenSer -> Client (audio: 88.99.3.10:60704) <-- Wrong port, MediaProxy set 60704 port !!! --------------------------------------------------------------------------- This is the mediaproxy/proxydispatcher log. As you can see media port open during INVITE is inmediatelly closed WHY?): --------------------------------------------------------------------------------------- ### 09:28:46 - 9:28:46 openser-host proxydispatcher[31874]: request 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net 192.168.1.203:10370:audio 88.99.1.192 sip.domain.net local 66.11.0.144 remote Domain=20-=20Asterisk=20PBX info=from:216@sip.domain.net,to:996511959@sip.domain.net,fromtag:as3c11216c,totag: - 09:28:46 openser-host mediaproxy[31869]: request 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net 192.168.1.203:10370:audio 88.99.1.192 sip.domain.net local 66.11.0.144 remote Domain=20-=20Asterisk=20PBX info=totag:,to:996511959@sip.domain.net,from:216@sip.domain.net,fromtag:as3c11216c - 09:28:46 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: started. listening on 88.99.3.10:60696 - 09:28:46 openser-host mediaproxy[31869]: execution time: 0.60 ms - 09:28:46 openser-host proxydispatcher[31874]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '88.99.3.10 60696' - 09:28:46 openser-host proxydispatcher[31874]: execution time: 2.45 ms **************** ERROR: Why does it occur ??? *********************** - 09:28:46 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) - 09:28:46 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: ended (did timeout). ********************************************************************* ### 09:28:48 - 09:28:48 openser-host proxydispatcher[31874]: lookup 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net 66.11.6.142:45734:audio 66.11.0.144 sip.domain.net local unknown unknown CS2000_NGSS/9.0 info=from:216@sip.domain.net,to:996511959@sip.domain.net,fromtag:as3c11216c,totag:4d764120338220087992847 - 09:28:48 openser-host mediaproxy[31869]: lookup 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net 66.11.6.142:45734:audio 66.11.0.144 sip.domain.net local unknown unknown CS2000_NGSS/9.0 info=totag:4d764120338220087992847,to:996511959@sip.domain.net,from:216@sip.domain.net,fromtag:as3c11216c - 09:28:48 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: started. listening on 88.99.3.10:60698 - 09:28:48 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: ended (did timeout). ### 09:28:58 - 09:28:58 openser-host proxydispatcher[31874]: lookup 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net 66.11.6.142:45734:audio 66.11.0.144 sip.domain.net local unknown unknown CS2000_NGSS/9.0 info=from:216@sip.domain.net,to:996511959@sip.domain.net,fromtag:as3c11216c,totag:4d764120338220087992847 - 09:28:58 openser-host mediaproxy[31869]: lookup 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net 66.11.6.142:45734:audio 66.11.0.144 sip.domain.net local unknown unknown CS2000_NGSS/9.0 info=totag:4d764120338220087992847,to:996511959@sip.domain.net,from:216@sip.domain.net,fromtag:as3c11216c - 09:28:58 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: started. listening on 88.99.3.10:60704 - 09:28:58 openser-host mediaproxy[31869]: execution time: 0.62 ms - 09:28:58 openser-host proxydispatcher[31874]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '88.99.3.10 60704' - 09:28:58 openser-host proxydispatcher[31874]: execution time: 1.15 ms **************** ERROR: Why does it occur ??? *********************** - 09:28:59 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) - 09:28:59 openser-host mediaproxy[31869]: session 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net: ended (did timeout). ********************************************************************* ### 09:29:05 - 09:29:05 openser-host proxydispatcher[31874]: delete 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net info= - 09:29:05 openser-host mediaproxy[31869]: delete 5be83da413f9dbd468a6d83818fb356d(a)sip.domain.net info= --------------------------------------------------------------------------------------- I also attach the complete SIP capture of this real example, but I see nothing wrong in it. The question is: Why MediaProxy ends the session during the INVITE? why does it close the port? Thanks a lot for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 1

[OpenSER-Users] Compiling / running Mediaproxy 2.0 on Debian Lenny
by jh＠tutpro.com 08 Jul '08

08 Jul '08

mediaproxy diapatcher works in ubuntu hardy, but not in lenny. the only difference i see is tls/python related versions is in python-gnutls: hardy: python-gnutls 1.1.4-1 lenny: python-gnutls 1.1.5-2 -- juha

2 4

Re: [OpenSER-Users] getting error while adding module to openser
by Bogdan-Andrei Iancu 08 Jul '08

08 Jul '08

Well, the error message is quite self explaining: *No such file or directory* check if the */usr/src/OpenSER/openser-1.2.0-tls/modules/pna.so *really exists. Regards, Bogdan suresh tattikota wrote: > > hi > i checked it in log file,i am getting these errors > > *ERROR: load_module: could not open module > </usr/src/OpenSER/openser-1.2.0-* > ** > *tls/modules/pna.so>: > /usr/src/OpenSER/openser-1.2.0-tls/modules/pna.so: cannot open shared > object file: No such file or directory > * > *openser: parse error (54,13-14): failed to load module* > > *ERROR: no version info in module > </usr/local/lib/openser/modules/pna.so>: > /usr/local/lib/openser/modules/pna.so: undefined symbol: module_version* > > * ERROR: no version info in module > </usr/local/lib/openser/modules/pna.so>: > /usr/local/lib/openser/modules/pna.so: undefined symbol: module_version > * > > > thank u > > > On 7/7/08, *Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi > > is tour openser starting or not? if not, look (eventually in > syslog) for any startup errors. > > Regards, > Bogdan > > suresh tattikota wrote: > > hi all > ** *this is the steps i followed to add module * > ** * i took one example module as exMod and this exMod sub > directory is in directory called /modules * > ** ** *1. created one makefile * > *2.one .c file with one printf statement , .h file for > corresponding .c file * > *3.and added loadmodule "exMod.so" in openser.cfg file then * > *4. i run the openser with * make prifix=/usr/local/ all and > make prifix=/usr/local/ install > 5 while i am running the openser it is not showing any thing > *the .c an .h file of my* *exMod are * > > *exMod.c:* > > void print_modname() > { > > printf("this is module name "); > } > > *exMod.h* > > void print_modname(); > > and my Makefile is : > > include ../../Makefile.defs > auto_gen= > NAME=exMod.so > DEFS+=-I$(LOCALBASE)/include > LIBS= > > include ../../Makefile.modules > > so please help me regarding this > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[OpenSER-Users] Strange REGISTER problem
by mark morreny 08 Jul '08

08 Jul '08

Hi, I am experiencing a very strange problem with REGISTER. If my SIP client registers to Openser with openser's internal IP ( 192.168.1.104), then it works. But if the SIP client registers using Openser's public IP, it does not work. The 200K returned from Openser is different for each case. For the 1st case, my SIP client recognizes the 200 OK, but for the 2nd case, my SIP client will keep on sending another REGISTER upon receiving the 200 OK. Here are the two different 200 OK msg that Openser sends back to the client. U 192.168.1.104:5060 -> 192.168.1.100:44522 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.100:44522 ;branch=z9hG4bK-d87543-7e47d706ec23b831-1--d87543-;rport=44522. To: "1006"<sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b23f. From: "1006"<sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104> >;tag=7e6bbb05. Call-ID: MGYyMjczODBmNTRkMzA3Y2JmZjk4ODM0MjkxOWI5ZDQ.. CSeq: 2 REGISTER. Contact: <sip:1006@192.168.1.100:16728;rinstance=d380362012deaf00>;expires=1909, <sip:1006@192.168.1.100:16727;rinstance=94567af350ff0cad>;expires=1870, <sip:1006@192.168.1.100:25572;rinstance=57a835af07252a9e>;expires=3286, < U 192.168.1.104:5060 -> 78.251.75.157:16156 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.100:2388 ;branch=z9hG4bK-d87543-f0701c5a1050361a-1--d87543-;rport=16156;received= 78.251.75.157. To: "1006"<sip:1006@58.251.75.157 <sip%3A1006(a)58.251.75.157> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b13f. From: "1006"<sip:1006@58.251.75.157 <sip%3A1006(a)58.251.75.157> >;tag=48233454. Call-ID: ZDVjODBlMjAxODQ5ZjAxYTRkYzRjYzUxMjY3YzAxNjk.. CSeq: 2 REGISTER. Contact: <sip:1006@192.168.1.100:16728;rinstance=d380362012deaf00>;expires=1493, <sip:1006@192.168.1.100:16727;rinstance=94567af350ff0cad>;expires=1454, <sip:1006@192.168.1.100:25572;rinstance=57a835af072 Any help will be greatly appreciated. Thanks, Mark

1 1

[OpenSER-Users] problem with linksys SPA 921
by Juan Backson 08 Jul '08

08 Jul '08

Hello I am using Linksys spa 921 to talk to openser. The problem I am having is that when the phone gets a 401 Unauthorized msg from Openser during REGISTER, instead of re-register with digest, it sends back another Unauthorized msg to openser. Does anyone know when this is happening? U 43.34.342.12:16309 -> 43.43.22.124:5060 REGISTER sip:43.34.342.12 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK-aed7e028. From: "1005" <sip:1005@43.34.342.12>;tag=b19e671c4f6697beo0. To: "1005" <sip:1005@43.34.342.12>. Call-ID: 6e601902-41d31f68(a)192.168.1.101. CSeq: 23942 REGISTER. Max-Forwards: 70. Contact: "1005" <sip:1005@192.168.1.101:5060>;expires=3600. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. . U 43.43.22.124:5060 -> 43.34.342.12:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5060 ;branch=z9hG4bK-aed7e028;received=43.34.342.12. From: "1005" <sip:1005@43.34.342.12>;tag=b19e671c4f6697beo0. To: "1005" <sip:1005@43.34.342.12 >;tag=329cfeaa6ded039da25ff8cbb8668bd2.60bb. Call-ID: 6e601902-41d31f68(a)192.168.1.101. CSeq: 23942 REGISTER. WWW-Authenticate: Digest realm="43.43.22.124", nonce="487395d9bbe456c575a7a45745b7ec760b372923". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 43.34.342.12:5060 -> 43.43.22.124:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5060 ;branch=z9hG4bK-aed7e028;received=43.34.342.12. From: "1005" <sip:1005@43.34.342.12>;tag=b19e671c4f6697beo0. To: "1005" <sip:1005@43.34.342.12 >;tag=329cfeaa6ded039da25ff8cbb8668bd2.60bb. Call-ID: 6e601902-41d31f68(a)192.168.1.101. CSeq: 23942 REGISTER. WWW-Authenticate: Digest realm="43.43.22.124", nonce="487395d9bbe456c575a7a45745b7ec760b372923". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 43.34.342.12:16309 -> 43.43.22.124:5060 REGISTER sip:43.34.342.12 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK-aed7e028. From: "1005" <sip:1005@43.34.342.12>;tag=b19e671c4f6697beo0. To: "1005" <sip:1005@43.34.342.12>. Call-ID: 6e601902-41d31f68(a)192.168.1.101. CSeq: 23942 REGISTER. Max-Forwards: 70. Contact: "1005" <sip:1005@192.168.1.101:5060>;expires=3600. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. . U 43.43.22.124:5060 -> 43.34.342.12:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5060 ;branch=z9hG4bK-aed7e028;received=43.34.342.12. From: "1005" <sip:1005@43.34.342.12>;tag=b19e671c4f6697beo0. To: "1005" <sip:1005@43.34.342.12 >;tag=329cfeaa6ded039da25ff8cbb8668bd2.60bb. Call-ID: 6e601902-41d31f68(a)192.168.1.101. CSeq: 23942 REGISTER. WWW-Authenticate: Digest realm="43.43.22.124", nonce="487395dd63fe51d4006fddeab235203120c8faca". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 43.34.342.12:5060 -> 43.43.22.124:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5060 ;branch=z9hG4bK-aed7e028;received=43.34.342.12. From: "1005" <sip:1005@43.34.342.12>;tag=b19e671c4f6697beo0. To: "1005" <sip:1005@43.34.342.12 >;tag=329cfeaa6ded039da25ff8cbb8668bd2.60bb. Call-ID: 6e601902-41d31f68(a)192.168.1.101. CSeq: 23942 REGISTER. WWW-Authenticate: Digest realm="43.43.22.124", nonce="487395dd63fe51d4006fddeab235203120c8faca". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . Any suggestion to solving this problem will be great. JB

3 2

[OpenSER-Users] Can't add user to Mysql DB
by Eliass Salim 08 Jul '08

08 Jul '08

I installed openser 1.3, now I loaded Mysql module and made the required changes in /etc/openser/openserctlrc file. When I run the command: openserctl add 100 100 100@localhost I get the follwoing error: ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user '192.168.10.182' to the database failed Any idea what's going on? thanks

1 0

[OpenSER-Users] regarding openser v 1.2.0 installation
by Naveen Kumar 08 Jul '08

08 Jul '08

Hi all, i deleted openser.pid from /var/runand then i started [root@info sbin]# openserctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded Starting OpenSER : ERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed and then i started like: [root@info sbin]# openser start 0(22725) cfg. warning: (28,15-16): tls support not compiled in 0(22725) cfg. warning: (29,10-13): tls support not compiled in 0(22725) cfg. warning: (30,21-22): tls support not compiled in 0(22725) cfg. warning: (31,21-22): tls support not compiled in 0(22725) cfg. warning: (32,34-35): tls support not compiled in 0(22725) cfg. warning: (33,14-19): tls support not compiled in 0(22725) cfg. warning: (34,20-21): tls support not compiled in 0(22725) cfg. warning: (35,20-21): tls support not compiled in 0(22725) cfg. warning: (36,16-17): tls support not compiled in 0(22725) loading module /usr/local/lib/openser/modules/mysql.so 0(22725) loading module /usr/local/lib/openser/modules/sl.so 0(22725) loading module /usr/local/lib/openser/modules/tm.so 0(22725) DEBUG: register_pv: tm 0(22725) xl_add_extra_spec: extra items list is not initialized 0(22725) xl_add_extra_spec: inserting extra item [T_branch_idx] at [0] 0(22725) xl_add_extra_spec: inserting extra item [T_reply_code] at [1] 0(22725) loading module /usr/local/lib/openser/modules/rr.so 0(22725) loading module /usr/local/lib/openser/modules/maxfwd.so 0(22725) loading module /usr/local/lib/openser/modules/usrloc.so 0(22725) loading module /usr/local/lib/openser/modules/registrar.so 0(22725) loading module /usr/local/lib/openser/modules/textops.so 0(22725) loading module /usr/local/lib/openser/modules/mi_fifo.so 0(22725) loading module /usr/local/lib/openser/modules/auth.so 0(22725) loading module /usr/local/lib/openser/modules/auth_db.so 0(22725) set_mod_param_regex: mi_fifo matches module mi_fifo 0(22725) set_mod_param_regex: found <fifo_name> in module mi_fifo [/usr/local/lib/openser/modules/] 0(22725) set_mod_param_regex: usrloc matches module usrloc 0(22725) set_mod_param_regex: found <db_mode> in module usrloc [/usr/local/lib/openser/modules/] 0(22725) set_mod_param_regex: usrloc matches module usrloc 0(22725) set_mod_param_regex: found <db_url> in module usrloc [/usr/local/lib/openser/modules/] 0(22725) set_mod_param_regex: auth_db matches module auth_db 0(22725) set_mod_param_regex: found <db_url> in module auth_db [/usr/local/lib/openser/modules/] 0(22725) set_mod_param_regex: auth_db matches module auth_db 0(22725) set_mod_param_regex: found <calculate_ha1> in module auth_db [/usr/local/lib/openser/modules/] 0(22725) set_mod_param_regex: auth_db matches module auth_db 0(22725) set_mod_param_regex: found <password_column> in module auth_db [/usr/local/lib/openser/modules/] 0(22725) set_mod_param_regex: rr matches module rr 0(22725) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <mf_process_maxfwd_header>(1) in module maxfwd [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <record_route>(0) in module rr [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <loose_route>(0) in module rr [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <append_hf>(1) in module textops [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <append_hf>(1) in module textops [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <www_authorize>(2) in module auth_db [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <www_challenge>(2) in module auth [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <save>(1) in module registrar [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <lookup>(1) in module registrar [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <append_hf>(1) in module textops [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <lookup>(1) in module registrar [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <append_hf>(1) in module textops [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <t_relay>(0) in module tm [/usr/local/lib/openser/modules/] 0(22725) find_cmd_export_t: found <sl_reply_error>(0) in module sl [/usr/local/lib/openser/modules/] ERROR: bad config file (9 errors) 0(22725) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy 0(22725) DEBUG: tm_shutdown : start 0(22725) DEBUG: tm_shutdown : emptying hash table 0(22725) DEBUG: tm_shutdown : releasing timers 0(22725) DEBUG: tm_shutdown : removing semaphores 0(22725) DEBUG: tm_shutdown : destroying tmcb lists 0(22725) DEBUG: tm_shutdown : done 0(22725) shm_mem_destroy please anybody help me. With Best Regards, Naveen.

5 7

[OpenSER-Users] ERROR: load_module: could not open module
by suresh tattikota 08 Jul '08

08 Jul '08

hi all i am getting these errors wile running the openser *ERROR: load_module: could not open module </usr/src/OpenSER/openser-1.2.0-* ** *tls/modules/pna.so>: /usr/src/OpenSER/openser-1.2.0-tls/modules/pna.so: cannot open shared object file: No such file or directory * *openser: parse error (54,13-14): failed to load module* *ERROR: no version info in module </usr/local/lib/openser/modules/pna.so>: /usr/local/lib/openser/modules/pna.so: undefined symbol: module_version* * ERROR: no version info in module </usr/local/lib/openser/modules/pna.so>: /usr/local/lib/openser/modules/pna.so: undefined symbol: module_version * thank u

2 1

[OpenSER-Users] help serMyAdmin centos 5
by Ricky Gutierrez 08 Jul '08

08 Jul '08

hi list, I have some problems installing serMyAdmin in centos 5 and openser 1.3.2, I have running tomcat5 and it loads very well, but when I add inside the file context.xml, it doesn't load , it show me a http STATUS 404.. that it can be happening? I show them my file context.xml <?xml version="1.0" encoding="UTF-8"?>  <Context>  <WatchedResource>WEB-INF/web.xml</WatchedResource>   </Context> <Context path="/serMyAdmin"> <Resource auth="Container" driverClassName="com.mysql.jdbc.Driver" maxActive="20" maxIdle=" 10" maxWait="-1" name="jdbc/openser_MySQL" type="javax.sql.DataSource" url="jdbc:mysql://loc alhost:3306/openser" username="sermyadmin" password="holacisco23/> </Context> Another point, the openser database and serMyAdmin don't export me, when I export the serMyAdmin openser.sql, he throws me the following error: ERROR 1054 (42S22) at line 6: Unknown column 'version' in 'field list' I waiting for their help, I congratulate to the serMyAdmin developer for this excellent application, he looks at very good to work with openser rickygm

1 0

[Serusers] Iptel's Documentation Resources (caution !! out of date )
by muhammad akl 07 Jul '08

07 Jul '08

well , today i was browsing the documentation section of the iptel.org to get more information about SER to include them in my graduation's presentation ( yea i'll get rid of that college :D ) what i notice there most of the documents are out of date and one cannot get full information about some issues ( the most interesting sentence was for the admin guide " warning outdated !! " ) so my question " is there any hope to see a well-organized documents that one can depends on , meaning rewrite some sections , wrting more howto's and the most important issue is to rewrite the INSTALL file that comes with the source code for SER " and for that i will make some documents especially directed for solaris10 and opensolaris' area soon some of them : 1- installing SER' requirements for solaris 10 and opensolaris 2- full explaination for SER compilation process on either solaris10 or opensolaris 3- install SAMP package ( apache2 , mysql5 , php5 ) either using coolstack or via package manager of opensolaris 4- installing serweb on either solaris 10 or opensolaris 5 integration of SER plus Asterisk and how to make the right configuration so if any one could help me in the preceding points or to help do more work like howto configure SEMS and SER for media streaming and other issues i'd appreciate that and finally sorry for my poor english :) BR muhammad

2 1

[OpenSER-Users] getting error while adding module to openser
by suresh tattikota 07 Jul '08

07 Jul '08

hi all ** *this is the steps i followed to add module * ** * i took one example module as exMod and this exMod sub directory is in directory called /modules * ** ** *1. created one makefile * *2.one .c file with one printf statement , .h file for corresponding .c file * *3.and added loadmodule "exMod.so" in openser.cfg file then * *4. i run the openser with * make prifix=/usr/local/ all and make prifix=/usr/local/ install 5 while i am running the openser it is not showing any thing *the .c an .h file of my* *exMod are * *exMod.c:* void print_modname() { printf("this is module name "); } *exMod.h* void print_modname(); and my Makefile is : include ../../Makefile.defs auto_gen= NAME=exMod.so DEFS+=-I$(LOCALBASE)/include LIBS= include ../../Makefile.modules so please help me regarding this

2 1

[OpenSER-Users] openser ported to Intel Debian 64bits?
by Pablo Bertuzzi 07 Jul '08

07 Jul '08

Hello Everyone!, I have to install openser in production these days. So, is there any plan to port openser to Debian 64bits on Intel machines? kind regards, Pablo.

2 1

[OpenSER-Users] DNS/ENUM resolution
by Yazid Hadj Said 07 Jul '08

07 Jul '08

Hi all, If I modify R-URI, the DNS/ENUM resolution (choice of destination of this request) is made with the original RURI or with the new RURI Regards Yazid

3 2

[OpenSER-Users] Can't add user to Mysql DB
by Eliass Salim 07 Jul '08

07 Jul '08

I installed openser 1.3, now I loaded Mysql module and made the required changes in /etc/openser/openserctlrc file. When I run the command: openserctl add 100 100 100@localhost I get the follwoing error: ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user '192.168.10.182' to the database failed Any idea what's going on? thanks

2 1

[OpenSER-Users] Crashes with OpenSER and RTP Proxy
by Samuel Muller 07 Jul '08

07 Jul '08

Hello all, This e-mail to find some help, i'm desperate. I've many stability problems, when using RTP Proxy + OpenSER. Maybe it's hardware, maybe ... I really don't know. -> Infos : Server : Dell PowerEdge dual Xeon, 2 GB RAM, PERC 6i RAID card. OS: i tried all these one : . ubuntu gutsy . ubuntu hardy . debian etch . debian lenny (actually running) the problems are the same on each distro. it's AMD64 architecture-based : Linux 2.6.18-6-amd64 #1 SMP Fri Jun 6 05:24:08 UTC 2008 x86_64 GNU/Linux I need mySQL, NAT, and Radius support (Radiator used). I compilated as indicated, to enable in the acc module the support of Radius. I used the libradius (radiusclient-ng 0.5.6), and tried too with the libradius-ng2. sources : http://www.openser.org/dokuwiki/doku.php/install:make-debian-packages http://www.openser.org/dokuwiki/doku.php/install:backport-debian-source-pac… and some related websites like voip-info.org, ... then, i tried these pairs : RTP Proxy 1.02 + OpenSER 1.3.1 : crash RTP Proxy 1.02 + OpenSER 1.3.2 : crash (actually running) RTP Proxy 1.1 + OpenSER 1.3.1 : crash RTP Proxy 1.1 + OpenSER 1.3.2 : crash But, at each first running, all works fine : NAThelper, Radius, mySQL. Registrations are ok, accounting tickets and mySQL requests too, calls are fine, ... And when i changed one line of config, for example, an AVP value, it crashes at restart or just at the start, for different reasons each time. The processes become ghosts, or never restart, etc... See all the debug files in attached files - in this case, it's the NAThelper who crashed - but sometimes it works really fine ... even if i don't touch any configuration. If you need more infos, or another kind of crash, no pbm ... Many thanks to all for any suggestions, this situation makes me crazy. Samuel MULLER

2 2

[OpenSER-Users] 2 problems with serMyAdmin
by Ahmed Huraimel 07 Jul '08

07 Jul '08

hello all, i am facing two problems with serMyAdmin.05-beta. the problems are 1- Error when adding a user in groups 2- wrong table attributes when adding user preferences. for the first problem, after i installed openser 1.3.2 with mySQL DB, i loaded the group model after connecting my openser to the PSTN. so i want to use the group model to permit the users for local, long distance and international calls. it work fine with me when i added the users using "openserctl acl add". but when i added the groups using serMyAdmin i get the folowing error message: *Message: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] Class: UserController At Line: [111] Code Snippet: org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:619) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at UserController$_closure6_closure11.doCall(UserController.groovy:111) at UserController$_closure6.doCall(UserController.groovy:101) at UserController$_closure6.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at $Proxy3.saveOrUpdate(Unknown Source) ... 32 more Caused by: com.mysql.jdbc.exceptions.jdbc4.MySQLIntegrityConstraintViolationException: Duplicate entry 'agentA-inrialpes.fr-' for key 2 at com.mysql.jdbc.Util.handleNewInstance(Util.java:406) at com.mysql.jdbc.Util.getInstance(Util.java:381) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1015) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 33 more* ========================================= the second problem is when adding a user references, basically i got a wrong attributes. specifically, the username is not stored in the database. see below the result *Username: agentA(a)inrialpes.fr Attribute: callfwd Value: sip:AgentB@inrialpes.fr <sip%3AAgentB(a)inrialpes.fr> Type: String:String Last Modified: 2008-07-07 17:45:02.0 *and in the MySQL database it is: +----+------+-------------+--------------+-----------+------+-------------------------+---------------------+---------+---------+ | id | uuid | username | domain | attribute | type | value | last_modified | version | user_id | +----+------+-------------+--------------+-----------+------+-------------------------+---------------------+---------+---------+ | 6 | 12 | 0 | inrialpes.fr | callfwd | 0 | sip:AgentB@inrialpes.fr <sip%3AAgentB(a)inrialpes.fr> | 2008-07-07 17:45:02 | 0 | 12 | +----+------+-------------+--------------+-----------+------+-------------------------+---------------------+---------+---------+ ========================================= does anyone faces thous problem? AHMED ALALI

1 0

[OpenSER-Users] Compiling / running Mediaproxy 2.0 on Debian Lenny
by jh＠tutpro.com 07 Jul '08

07 Jul '08

do you use gnutls 2.4.0? It seems that they renamed/ obseleted this function there [1]: gnutls_openpgp_crt_get_key_id: ADDED, obsoletes gnutls_openpgp_crt_get_id. dan, any chance to that you would update dispatcher to be compatible with gnutls 2.4.1? -- juha

1 0

[OpenSER-Users] Problem with multiple openser instances and attended transfers (REFER with replaces part)
by Ardjan Zwartjes 07 Jul '08

07 Jul '08

Hi all, I'm currently working on a VoIP system with multiple opensers as registrars, clients can register to whatever openser they like and use that one as an outgoing proxy. The configuration scripts makes sure that invites are branched to all other opensers so that the call is delivered wherever a client is registered. Basic call functionality is working fine, attended transfers however are a different story. If all the clients are registered on the same openser this works perfectly. Client A calls client B, client B calls client C, B sends a REFER to A with a replaces in the Refer-To header and A calls client C with a Replaces header in the INVITE. The INVITE from A to C here goes through the openser since all clients use the openser as a outgoing proxy and all is well. In the setup with multiple opensers however the problem arises. If client C is registered on a different openser then A and B the third call won't go through the registrar for C, the proxy for A and B will try to contact C directly since there is no route information in a REFER on how client C can be reached. If C is behind a firewall or the registrar for C knows specific (network) routes on how to contact C this scenario will fail since the INVITE from the openser for A and B directly to client C won't arrive. So my question is: Does anybody has any ideas on how to reach the desired goal: The INVITE from A to C should go through the openser for A and the openser for C. Personally I was thinking in the line of a reverse registration lookup to check if a contact is known on an openser and if it isn't I could branch it to all other openser systems, but I couldn't find a function that does this reverse lookup. I could try to implement this function myself, but I would prefer existing (tested ;)) functionality. Does anybody know a function that does this, or can anybody think of another solution for this problem? Thanks in advance. Kind regards, Ardjan Zwartjes.

1 0

[OpenSER-Users] Compiling / running Mediaproxy 2.0 on Debian Lenny
by Thomas Gelf 07 Jul '08

07 Jul '08

Hi list, I would like to give Mediaproxy 2.0 a try - it compiles fine on Debian Lenny (dpkg-buildpackage). However starting it results in "fatal error: failed to create MediaProxy Relay: /usr/lib/libgnutls-extra.so.26: undefined symbol: gnutls_openpgp_crt_get_id". Could anyone give me a little hint on how to resolve this issue? Named file /usr/lib/libgnutls-extra.so.26 is part of Lenny's libgnutls26, System is running semi-virtualized in a linux-vserver guest (64bit) - host system is running kernel 2.6.22.18. Kind regards, Thomas Gelf

2 1

[OpenSER-Users] using voipswitch as media server
by Tushar 07 Jul '08

07 Jul '08

Hi OPENSER community, I have recently started playing around with Openser, I am looking for a solution in which my Openser Server should carry the signalling part and my media stream should be handle by my voipswitch so I can monitor my RTP stream, work with my calling card solution. While browsing I came to know about mediaproxy rely and media server may be it help me to sent my RTP packet from my server but I like to use my voip switch for my media stream. Thanks, any input appreciated! Tushar Jain

4 7

[OpenSER-Users] Need Help on RLS.
by Mahesh Peddi 07 Jul '08

07 Jul '08

Hi Experts, I was working on open SER. And i need to test the RLS feature. I am approaching like this first i install openSER1.3.0 and then will deploy RLS rpm of which i got from the link http://fedora.nad.go.id/updates/8/i386/repodata/repoview/openser-rls-0-1.3.… . Could you suggest any other approach? Where i can get the working OpenSER and RLS? Thanks Regards, Mahesh Peddi

2 1

[OpenSER-Users] Nathelper tables
by Diego Costantini 07 Jul '08

07 Jul '08

Hi, I was following this sample configuration for a multihomed server: http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy and it mentions: "For this example to work, you must also follow the nathelper module instructions for creating new location_internal and location_external tables. Be careful to use underscores ("_") and not hyphens ("-") when creating the new tables." But I couldn't find any instruction anywhere. Can someone address me to the correct place please? Thanks, Diego

1 0

[OpenSER-Users] how to add new module
by suresh tattikota 07 Jul '08

07 Jul '08

hi all please suggest me how to add new module to openser now i am using openser 1.2 please guide me how to procede ..... thank u in advance

2 1

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