- sr-users - kamailio.org

[Users] dialog expired accounting
by tele 24 Oct '06

24 Oct '06

Hi all, I've read about the dialog module and the new "default_timeout" or "timeout_avp" features. What i need to know it's if in case of default_timeout or timeout by avp the calls are anyway accounted. How are accounted the calls if dialog timeout expired? thanks :tele

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[Users] OpenSER Dokuwiki updated
by Daniel-Constantin Mierla 24 Oct '06

24 Oct '06

Hello, for those which haven't spotted yet, OpenSER dokuwiki site has been updated. Lot of useful documentation was brought to light, the new index should ease the browsing and let find quickly the relevant pages. Also, the information is grouped now on namespaces, based on topics. We would like to thank to all that contributed with examples, code snippets or they shared the VoIP experiences there. http://www.openser.org/dokuwiki/ Cheers, Daniel

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[Users] OpenSER Summit - Preview (I)
by Daniel-Constantin Mierla 24 Oct '06

24 Oct '06

Hello, the agenda is getting close to final version. I just want to preview the content of the event, now with two weeks before the event. I believe we manged to bring there in discussion most of the domains VoIP could touch and hot topics nowadays. We cover from development to business cases, from research to IMS, 3/4G. Next is a briefing of agenda, as of today, although it may change a bit in the future. The summit day (November 8) will start with introduction of special guests and history of OpenSER. It will be followed by a set of heavy subjects of VoIP: Application Server based on Java SipServlet technology, geographic redundancy and VoIP on mobile network. VoIPUser open VoIP services will then show their statistics and evolution over the years, and how they become one of the biggest VoIP communities world wide. Noon break will allow to visit the exhibition and group chats for hot discussions. Afternoon session will bring very interesting subjects: safe peering; high availability; OpenSER IMS prototype; developing VoIP applications; a look educational VoIP infrastructures - INRIA & SIP.edu plus a walk through VoIP opportunities. We end the day with what should be very entertaining: open discussion about OpenSER future and roadmap. This is not all, there are other special guests that can give you key notes about other topics and products of VoIP. As usual, details at: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… More will come as the event approaches. Cheers, Daniel

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[Users] X-lite is not Working, But SipEyeMedia Softphones is Working Fine.
by sunkara 24 Oct '06

24 Oct '06

Hello Users, I don't Know What Happening in this Issues... Help me in this . When Testing the X-lite Softphones, is not working. When I use the Sip EyeMedia Softphone it Working Fine. Is Problem in X-lite Configuration or Router(Firewall Gateway Keepers ) is blocking it... Help me . -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */

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[Serusers] Dropping Replies
by Sebastian Damm 24 Oct '06

24 Oct '06

Hi, I know that what I want to do is a bit odd but here is why: When using Asterisk as PSTN gateway and using SIP INFO for DTMF digits, there is a problem with devices not understanding SIP INFO. When Asterisk sends out its INFO packet and gets back a negative reply, it stops bridging and throws away the whole call. That is a really bad issue on Asterisk side, but I just wanted to find a workaround. So what I wanted to do is drop negative replies on INFO packets, so Asterisk just retransmits them and does not get any feedback. Those INFO packets come through my loose route and there I put the following little statement in there: if(method=="INFO" && src_ip == 10.0.0.1 ) { xlog("L_NOTICE"," - INFO Request: %mb"); t_on_reply("7"); setflag(3); }; With that answers jump into onreply_route[7]. This part works ok. onreply_route[7] { if (search("CSeq:[0-9 ]+INFO")) { xlog("L_NOTICE"," - INFO Reply: %mb"); if(t_check_status("405")) { xlog("L_NOTICE"," - Dropping INFO Reply"); drop(); } } } This part does not work. I also tried creating a route just for dropping and then jumpinng into this route from my reply_route, but then I also only have the log line from that route, but drop does not work. I tried sl_send_reply, t_release and other commands of which I thought they might be helpful. But the "offending answer" is always transmitted to my gateway. Does anybody have an idea how to just "throw away" reply packets? Thanks in advance, Sebastian

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[Users] generating ACK
by Ramin Dousti 23 Oct '06

23 Oct '06

Hello, How can I generate an ACK locally on openser proxy as a responsetof a failure? And subsequently drop the failed message. In other word I do not want to pass the failed message back to my UA, I just want to process it on openser... Thanks in advance, -- Ramin

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[Users] Changing TM Retransmission Count?
by Tavis P 23 Oct '06

23 Oct '06

Reading through the documentation for the TM module (1.1.0) i've been unable to find anything indicating that i can change the retransmission count, only the retransmission intervals By my count OpenSER is retransmitting an INVITE to an endpoint user 12 times and i would like to reduce this as it is interfering with the call forward no answer logic i've put in place (timer doesn't hit until all 12 retransmissions are sent, which is about 42 seconds using the default timers) Do i have any options other than modifying the tm module code? If so, could someone give me a pointer where i need to look (i've had a quick read and its rather complicated so i might take me some time to find the logic i'm looking for) tavis

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[Users] aliases
by Benko 23 Oct '06

23 Oct '06

Hello! Are entries in the aliases table only read at startup? While a record that was in the table before i started openser is available since the beginning, a entry i inserted a while ago(+1h) but after starting openser is still not working(and also doesn't appear in the memory cache). I'd be glad if someone can give me info on this particular matter, thx! Regards Christian p.s.: just stumbled over dbaliases, guess this will work better...

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[Users] Enabling caller id block on outbound calls.
by Shane Burrell 23 Oct '06

23 Oct '06

We have grandstreams and a few customers that want to block there outbound caller id. Has anyone done this? I'd really like the *30 and *31 feature to work but if you turn it on the openser server blocks the call. TIA, Shane

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[Serusers] Problem with radius - ser setup
by Lokesh Kumar 23 Oct '06

23 Oct '06

Hello List , I have problem with the authentication of users via Radius. Mentioned below are the logs. Can anyone please give me hint where I am doing wrong , I did exactly as mentioned in the ser-radius how to. The logs from Radius are like the mentioned below. Sip-Uri-User = "211069020" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 14421 modcall[authorize]: module "preprocess" returns ok for request 14421 modcall[authorize]: module "chap" returns noop for request 14421 modcall[authorize]: module "mschap" returns noop for request 14421 rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 14421 rlm_realm: Looking up realm "xxx.pt" for User-Name = "211069020(a)XXX.pt" rlm_realm: No such realm "xxx.pt" modcall[authorize]: module "suffix" returns noop for request 14421 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 14421 users: Matched entry DEFAULT at line 152 users: Matched entry 211069020(a)xxx.pt at line 217 modcall[authorize]: module "files" returns ok for request 14421 modcall: leaving group authorize (returns ok) for request 14421 rad_check_password: Found Auth-Type Digest auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 14421 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "211069020" Digest-Realm = "xxx.pt" Digest-Nonce = "453c9377946262d76fceca014a1553f8384db20f" Digest-URI = "sip:xxx.pt" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "00000001" Digest-CNonce = "5640622967614" A1 = 211069020:xxx.pt:211069020 A2 = REGISTER:sip:xxx.pt KD = f7d0e83a9277bd217ba41ac8e070aee4:453c9377946262d76fceca014a1553f8384db20f:00 000001:5640622967614:auth:4a210dfe3dc88ca825764f5ea20d8b01 modcall[authenticate]: module "digest" returns ok for request 14421 modcall: leaving group authenticate (returns ok) for request 14421 radius_xlat: 'Authenticated from Radius' Sending Access-Accept of id 131 to 127.0.0.1 port 56964 Reply-Message = "Authenticated from Radius" Sip-Rpid = "211069020" Finished request 14421 Going to the next request --- Walking the entire request list --- Waking up in 6 seconds... rad_recv: Access-Request packet from host 127.0.0.1:56965, id=132, length=57 User-Name = "@xxx.pt" Service-Type = Callback-Administrative NAS-Port = 0 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 14422 modcall[authorize]: module "preprocess" returns ok for request 14422 modcall[authorize]: module "chap" returns noop for request 14422 modcall[authorize]: module "mschap" returns noop for request 14422 modcall[authorize]: module "digest" returns noop for request 14422 rlm_realm: Looking up realm "xxx.pt" for User-Name = "@xxx.pt" rlm_realm: No such realm "xxx.pt" modcall[authorize]: module "suffix" returns noop for request 14422 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 14422 users: Matched entry DEFAULT at line 152 modcall[authorize]: module "files" returns ok for request 14422 modcall: leaving group authorize (returns ok) for request 14422 rad_check_password: Found Auth-Type System auth: type "System" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 14422 rlm_unix: Attribute "User-Password" is required for authentication. modcall[authenticate]: module "unix" returns invalid for request 14422 modcall: leaving group authenticate (returns invalid) for request 14422 auth: Failed to validate the user. Delaying request 14422 for 1 seconds Finished request 14422 Going to the next request Waking up in 6 seconds... --- Walking the entire request list --- Cleaning up request 14421 ID 131 with timestamp 453c9257 Sending Access-Reject of id 132 to 127.0.0.1 port 56965 Cleaning up request 14422 ID 132 with timestamp 453c9257 Nothing to do. Sleeping until we see a request. And the logs from ser are like the mentioned below. 0(15034) found end of header 0(15034) find_first_route: No Route headers found 0(15034) loose_route: There is no Route HF 0(15034) parse_headers: flags=-1 0(15034) check_via_address(212.13.42.65, 192.168.0.47, 0) 0(15034) parse_headers: flags=64 0(15034) check_nonce(): comparing [453c93b9459779f9f51440d01f13c9e0db2b2965] an d [453c93b9459779f9f51440d01f13c9e0db2b2965] 0(15034) radius_authorize_sterman(): Success 0(15034) save_rpid(): rpid value is '211069020' 0(15034) radius_does_uri_exist(): Failure 0(15034) parse_headers: flags=-1 0(15034) check_via_address(212.13.42.65, 192.168.0.47, 0) 0(15034) DEBUG:destroy_avp_list: destroying list 0xf5062350 0(15034) receive_msg: cleaning up 0(15034) SIP Request: 0(15034) method: <OPTIONS> 0(15034) uri: <sip:xxx.pt:5060> 0(15034) version: <SIP/2.0> 0(15034) parse_headers: flags=1 0(15034) Found param type 235, <rport> = <n/a>; state=6 0(15034) Found param type 232, <branch> = <z9hG4bKc0a8002f0000000b453c94d700006 3800000fc1b>; state=16 0(15034) end of header reached, state=5 0(15034) parse_headers: Via found, flags=1 0(15034) parse_headers: this is the first via 0(15034) After parse_msg... 0(15034) preparing to run routing scripts... 0(15034) parse_headers: flags=128 0(15034) DEBUG: get_hdr_body : content_length=0 0(15034) get_hdr_field: cseq <CSeq>: <11858> <OPTIONS> 0(15034) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(15034) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(15034) DEBUG: add_param: tag=56414607811795 0(15034) end of header reached, state=29 0(15034) parse_headers: flags=256 0(15034) end of header reached, state=9 0(15034) DEBUG: get_hdr_field: <To> [29]; uri=[sip:xxx.pt:5060] 0(15034) DEBUG: to body [<sip:xxx.pt:5060> ] 0(15034) found end of header 0(15034) find_first_route: No Route headers found 0(15034) loose_route: There is no Route HF 0(15034) lookup(): '' Not found in usrloc 0(15034) lookup(): '' Not found in usrloc 0(15034) parse_headers: flags=-1 0(15034) check_via_address(212.13.42.65, 192.168.0.47, 0) 0(15034) DEBUG:destroy_avp_list: destroying list (nil) 0(15034) receive_msg: cleaning up The users file in raddb is like this 211069020(a)xxx.pt Auth-Type := Digest, User-Password == "211069020" Reply-Message = "Authenticated from Radius", Sip-Rpid = "211069020" 211069020(a)xxx.pt Sip-Group == "local", Auth-Type := Accept Reply-Message = "Authorized" "users" 221L, 7200C I have mentioned dictionary.ser in radius and radiusclient. Where I am doing wrong can anyone please give a hint. Thank you very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.9/490 - Release Date: 10/20/2006

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[Serusers] ser 0.10 status
by Mark Price 23 Oct '06

23 Oct '06

How far is ser 0.10 from being usable in production? mark

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[Serusers] SER + RTP Proxy Silence
by Abdul Qadir 23 Oct '06

23 Oct '06

Hi, I m running SER 0.9.6 on FC5. with rtpproxy running on same machine. The communicatin betwwn SER and rtpproxy is 100 % ok since when ever I make calls from one softphone to another I get valid ports,plus i also confirmed by the messages displayed by rtpproxy like new command etc and other mappings. The problem is rtpproxy is not relaying the RTP packets, although I confirmed the RTP packets from both clients reached the SER + rtpproxy machine and are sent on the same port specified by rtpproxy. Can any body tell whats wrong Best Regards, Abdul Qadir --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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[Serusers] nat + MESSAGE packets
by Mark Price 23 Oct '06

23 Oct '06

Hi, I'm finding that my ser clients can't can only unreliably communicate using MESSAGE packets, and that presence and presence subscriptions are equally unreliable (i.e. very). The odd thing is that they do sometimes, in fact, work. Anyways, I discovered that there are definitely NAT issues going on, since I can see inbound MESSAGE packets banging on the outside of my firewall. What are people doing to mitigate this? Mark

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[Serusers] How to use domain module without database or with radius???
by Lokesh Kumar 23 Oct '06

23 Oct '06

Hi list, I have an issue with domain module which needs database support, and I want to use radius, but in domain module there is no mention of using this module without database, how to solve this problem please suggest something Thanks you very much Regards Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.5/483 - Release Date: 10/18/2006

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[Serusers] rfc 3262 support
by Kamal.Mann＠t-systems.com 23 Oct '06

23 Oct '06

Hi All Do SER supports 'rfc 3262 Reliability of Provisional Responses in SIP'? Is there any list of standards supported by SER available?? Regards Kamal Mann

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[Serusers] reinvite accounting
by olivier.taylor 23 Oct '06

23 Oct '06

hi all, Using Ser 0.9.6 and asterisk(s) as pstn gateways. Asterisks are used for the Billing on the pstn. For sip --> sip calls, no billing, just accounting of the normal and missed calls. The problem is REINVITEs, ACC module record all the Invites and ACK, even when it's a Reinvite (has_totag). I use the flags for telling Ser what to account, I never set the flags for Reinvites, not even reset the flags. Is ther a simple way to avoid recording of Reinvites in the ACC table? Thanks, Olivier

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[Serusers] Re: pa module: presence "substatus"
by Vaclav Kubart 23 Oct '06

23 Oct '06

Hi, you have to set LD_LIBRARY_PATH variable before running ser (see "3.4 Running SER" in presence handbook): export LD_LIBRARY_PATH=/base/ser/directory/lib/ser /base/ser/directory/sbin/ser -f /base/ser/directory/etc/ser/ser.cfg Vaclav On Fri, Oct 20, 2006 at 07:06:54PM +0200, Diego Do?ate wrote: > > Hi, > > I have done followed these steps, but still the same errors. Should I > unistall something else? > ... > 0(13168) qm_malloc(0x8113ea0, 24) returns address 0x8148710 frag. 0x81486f8 > (size=24) on 1 -th hit > 0(13168) read 2739142786 from /dev/urandom > 0(13168) seeding PRNG with 3900519771 > 0(13168) test random number 509866974 > 0(13168) ERROR: load_module: could not open module > </usr/local//lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open shared > obj ect file: No such file or directory > 0(13168) parse error (36,13-14): failed to load module > 0(13168) ERROR: load_module: could not open module > </usr/local//lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared > objec t file: No such file or directory > 0(13168) parse error (37,13-14): failed to load module > 0(13168) ERROR: load_module: could not open module > </usr/local//lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared > object file: No such file or directory > 0(13168) parse error (38,13-14): failed to load module > 0(13168) ERROR: load_module: could not open module > </usr/local//lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot open > sha red object file: No such file or directory > 0(13168) parse error (39,13-14): failed to load module > 0(13168) set_mod_param_regex: No module matching <rls> found > 0(13168) parse error (76,21-22): Can't set module parameter > 0(13168) set_mod_param_regex: No module matching <rls> found > 0 > .... > > > Diego > > -----Mensaje original----- > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Enviado el: viernes, 20 de octubre de 2006 7:15 > Para: Diego Do?ate > CC: serdev-bounces(a)lists.iptel.org > Asunto: Re: pa module: presence "substatus" > > Hi, > this seems that you are compiling something else than "presence snapshot" > because "presence snapshot" has modified Makefiles to compile libraries > automaticaly which was not done in your case. > > Try to follow the compilation rules "3.2 Installation from CVS": > > cd sip_router/lib > make -f Makefile.ser install prefix=/base/ser/directory cd .. > make install include_modules="dialog pa rls presence_b2b xcap mysql xmlrpc" > prefix=/base/ser/directory > > [in your case will be prefix=/usr/local if you want to install ser there] > > Vaclav > > On Thu, Oct 19, 2006 at 07:57:55PM +0200, Diego Do?ate wrote: > > > > Hi, > > > > Thanks for the suggestion. I have download the 0.10.99 version > > ("presence > > snapshot") but I can not make it work: after the "make install", I go > > to /usr/local/etc/ser, where is the ser.cfg to run, but with ser -f > > ser.cfg it gives me this errors: > > > > ERROR: bad config file (47 errors) > > -bash-3.00# ser -f ser.cfg -E > > 0(15125) ERROR: load_module: could not open module > > </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open > > shared object file: No such file or directory > > 0(15125) parse error (36,13-14): failed to load module > > 0(15125) ERROR: load_module: could not open module > > </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open > > shared object file: No such file or directory > > 0(15125) parse error (37,13-14): failed to load module > > 0(15125) ERROR: load_module: could not open module > > </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open > > shared object file: No such file or directory > > 0(15125) parse error (38,13-14): failed to load module > > 0(15125) ERROR: load_module: could not open module > > </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot > > open shared object file: No such file or directory > > 0(15125) parse error (39,13-14): failed to load module > > 0(15125) set_mod_param_regex: No module matching <rls> found > > > > > > And the last line is repeated with the 4 modules it can not find. But > > the 4 of them exists in /usr/local/lib/ser/modules, where it is looking > for... > > > > I thought the installation was going to be easy, as read in > > http://ftp.iptel.org/pub/ser/presence/presence-handbook/ar01s03.html (3.3. > > Installation from presence-snapshot), but there is something I am > > doing wrong. > > > > Any suggestion? I attach the .cfg > > > > Thanks a lot > > > > Diego Do?ate > > > > > > -----Mensaje original----- > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > miércoles, 18 de octubre de 2006 11:22 > > Para: Diego Do?ate > > CC: serdev-bounces(a)lists.iptel.org > > Asunto: Re: pa module: presence "substatus" > > > > Hi, > > I'm afraid that there was not good support for presence in 0.9.6. I > > recommend to use "presence snapshot". > > Vaclav > > > > On Wed, Oct 18, 2006 at 10:37:27AM +0200, Diego Do?ate wrote: > > > > > > Hi, > > > > > > I am using the last version of the SER server (0.9.6), and have > > > a problem with the presence management. I want a user (805) to > > > subscribe to the presence of another user (806), but when 805 sends > > > a SUBSCRIBE to the SER, it responses a NOTIFY with "closed-offline", > > > while both users had sent previously a successful REGISTER to the SER. > > > > > > Both users are added (serctl) to the server, and are displayed > > > in "subscriber" table of the "ser" DB. Once registered, they are > > > also in the "location" table. But the ser does not add "open" to the > > > NOTIFY.. Any > > idea? > > > > > > Once the user 805 is successfully subscribed to the presence of > > > user 806, when 806 modifies its presence (sends a PUBLISH to the > > > SER, with > > "open" > > > and <note>"Away"</note> in the xml, for instance), the SER is > > > suppposed to send a NOTIFY with the same xml to user 805 > > > (subscriber), > > isn't it? > > > > > > "pa" module is added, but in this moment I don't need > > > authentication. U can see the ser.cfg file bellow: > > > > > > Thanks in advance > > > > > > > > > // ******************************** > > > # > > > # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # > > > simple quick-start config script # > > > > > > # ----------- global configuration parameters > > > ------------------------ > > > > > > #debug=3 # debug level (cmd line: -dddddddddd) > > > #fork=yes > > > #log_stderror=no # (cmd line: -E) > > > > > > /* Uncomment these lines to enter debugging mode fork=no > > > log_stderror=yes */ > > > > > > check_via=no # (cmd. line: -v) > > > dns=no # (cmd. line: -r) > > > rev_dns=no # (cmd. line: -R) > > > #port=5060 > > > #children=4 > > > fifo="/tmp/ser_fifo" > > > > > > # ------------------ module loading > > > ---------------------------------- > > > > > > # Uncomment this if you want to use SQL database loadmodule > > > "/usr/local/lib/ser/modules/mysql.so" > > > > > > loadmodule "/usr/local/lib/ser/modules/sl.so" > > > loadmodule "/usr/local/lib/ser/modules/tm.so" > > > loadmodule "/usr/local/lib/ser/modules/rr.so" > > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > > > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > > > loadmodule "/usr/local/lib/ser/modules/registrar.so" > > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > > > > > loadmodule "/usr/local/lib/ser/modules/pa.so" > > > > > > # Uncomment this if you want digest authentication # mysql.so must > > > be loaded ! > > > #loadmodule "/usr/local/lib/ser/modules/auth.so" > > > #loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > > > > > # ----------------- setting module-specific parameters > > > --------------- > > > > > > # -- usrloc params -- > > > > > > #modparam("usrloc", "db_mode", 0) > > > > > > # Uncomment this if you want to use SQL database # for persistent > > > storage and comment the previous line modparam("usrloc", "db_mode", > > > 2) > > > modparam("pa","db_url","mysql://ser:heslo@localhost/ser") > > > > > > # -- auth params -- > > > # Uncomment if you are using auth module # #modparam("auth_db", > > > "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to > > > yes (which true in this config), # uncomment also the following > > > parameter) # #modparam("auth_db", "password_column", "password") > > > > > > # -- rr params -- > > > # add value to ;lr param to make some broken UAs happy > > > #modparam("rr", "enable_full_lr", 1) > > > > > > # ------------------------- request routing logic > > > ------------------- > > > > > > # main routing logic > > > > > > route{ > > > > > > # initial sanity checks -- messages with > > > # max_forwards==0, or excessively long requests > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483","Too Many Hops"); > > > break; > > > }; > > > if (msg:len >= 2048 ) { > > > sl_send_reply("513", "Message too big"); > > > break; > > > }; > > > > > > > > > # we record-route all messages -- to make sure that > > > # subsequent messages will go through our proxy; that's > > > # particularly good if upstream and downstream entities > > > # use different transport protocol > > > if (!method=="REGISTER") record_route(); > > > > > > # subsequent messages withing a dialog should take the > > > # path determined by record-routing > > > if (loose_route()) { > > > # mark routing logic in request > > > append_hf("P-hint: rr-enforced\r\n"); > > > route(1); > > > break; > > > }; > > > > > > if (!uri==myself) { > > > # mark routing logic in request > > > append_hf("P-hint: outbound\r\n"); > > > route(1); > > > break; > > > }; > > > > > > # if the request is for other domain use UsrLoc > > > # (in case, it does not work, use the following command > > > # with proper names and addresses in it) > > > if (uri==myself) { > > > > > > if (method=="PUBLISH"){ > > > if(t_newtran()){ > > > handle_publish("registrar"); > > > break; > > > } > > > } > > > if (method=="SUBSCRIBE") { > > > if (!t_newtran()) { > > > log(1, "newtran error\n"); > > > sl_reply_error(); > > > }; > > > handle_subscription("registrar"); > > > break; > > > }; > > > > > > if (method=="REGISTER") { > > > > > > # Uncomment this if you want to use digest authentication > > > # if (!www_authorize("192.168.1.118", "subscriber")) { > > > # www_challenge("192.168.1.118", "0"); > > > # break; > > > # }; > > > > > > save("location"); > > > break; > > > }; > > > > > > lookup("aliases"); > > > if (!uri==myself) { > > > append_hf("P-hint: outbound alias\r\n"); > > > route(1); > > > break; > > > }; > > > > > > # native SIP destinations are handled using our USRLOC DB > > > if (!lookup("location")) { > > > sl_send_reply("404", "Not Found"); > > > break; > > > }; > > > }; > > > append_hf("P-hint: usrloc applied\r\n"); > > > route(1); > > > } > > > > > > route[1] > > > { > > > # send it out now; use stateful forwarding as it works reliably > > > # even for UDP2TCP > > > if (!t_relay()) { > > > sl_reply_error(); > > > }; > > > } > > > > > > > > > > > > > > > > > > // ******************************** > > > > > > -----Mensaje original----- > > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > > miércoles, 18 de octubre de 2006 6:21 > > > Para: Diego Do?ate > > > CC: serusers(a)iptel.org > > > Asunto: Re: [Serdev] pa module: presence "substatus" > > > > > > Hi, > > > notes are quite often used for this. > > > > > > Can you send your config here? What SER version do you use? And if > > > you try register your users via sending REGISTER messages it is > > > still the > > same? > > > > > > Vaclav > > > > > > P.S. Send these messages to serusers too, might be that somebody > > > else will have the same problem... > > > > > > On Tue, Oct 17, 2006 at 05:18:55PM +0200, Diego Do?ate wrote: > > > > > > > > Hi, > > > > > > > > Thanks, then I will use a <note> tag for the presence > > > > substatus(<note>Away<\note>, for instance). > > > > > > > > I have a problem with the Subscription. User A subscribes to > > > > the presence of user B and receives a NOTIFY but with > > > > "close/offline", and both users are previously registered successfully > in SER. > > > > > > > > Both users are added with serctl, but it seems the SER does not > > "save" > > > > user A registration when user B subscribes... Where may be the > > > > problem? May it be related to autherization? By now, it is disabled... > > > > > > > > Thanks in advance > > > > > > > > > > > > > > > > -----Mensaje original----- > > > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > > > lunes, > > > > 16 de octubre de 2006 18:57 > > > > Para: Diego Do?ate > > > > CC: serdev(a)lists.iptel.org > > > > Asunto: Re: [Serdev] pa module: presence "substatus" > > > > > > > > Hi, > > > > support for PIDF extensions (elements from non-pidf namespace) is > > > > nearly finished in current CVS version, but it still needs some work. > > > > It will be done as soon as possible. > > > > > > > > Notes should be possible in last presence snapshot without any > > > > problems > > > > - they can be within tuple or presence elements according > specification. > > > > > > > > Vaclav > > > > > > > > On Mon, Oct 16, 2006 at 05:52:47PM +0200, Diego Do?ate wrote: > > > > > > > > > > Hi, > > > > > > > > > > In the SER "pa" module, I would like to use a tag like "note" > > > > > or "substatus" in the pidf of the NOTIFY when a PUBLISH is > > > > > received (with a presence change, to "Away", for instance), but > > > > > SER only modified the <status> tag ("open"/"closed"). > > > > > > > > > > How can I implement a more detailed presence status > > > > > management in the SER? > > > > > > > > > > Thanks in advance > > > > > > > > > > > > > > > _______________________________________________ > > > > > Serdev mailing list > > > > > Serdev(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serdev > > > > > > > > > >

1 0

[Serusers] Windows Messenger 5.0 error. Please help!!
by Meidiana Ten 23 Oct '06

23 Oct '06

Hi all, I'm using Windows Messenger 5.0 as SIP communication to send IM between clients. It has successfully log into SER server I've built. But when another client log in, the icon on messenger doesn't change to online.I try to send IM but It failed, but when I call using x-lite, it works. What's the problem. Please help!! Regards, Meidiana --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

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[Serusers] Presence Server Question
by Dave 22 Oct '06

22 Oct '06

Hello, I have two SER servers - One acting as a proxy (0.9.6) and the second as a presence (0.10). I am forwarding SUBSCRIBE, MESSAGE, PUBLISH requests from proxy to Presnece server. How can I make the presence server know of "who is online"? Do I need the presence server to point to the "location" table of the Proxy Server or do I need to forward REGISTERS from proxy to Presence server?? Any help will be much appreciated __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] TLS: has TLS made communication secure? Please help...
by Ferianto siregar 21 Oct '06

21 Oct '06

Dear all, Thank you very much for this chance. Thank you very much for your time to read my problem. Thanks All, I still get some doubts in understanding TLS because untill now, my lecturer still want me to evidence that the communication with TLS support is secure eventhough I have told him that the securing communication in TLS is in the encyption method. But, He still does not believe me. In his opinion, He said that I have to evidence it by eavesdropping the communication, so I can show the difference between communication with TLS and the communication without TLS. Of course, I am confused because as I know frOm the mailinglist and TLS tutorial, it is not necessary to proof because the security is in encryption in TLS. Does anybody can help me to solve my problem? Here are some question that I have to ask to you. Maybe the answer that I got from these questions, can I use to evidence the secure communication in TLS. Please help me..Please 1. Would anybody mind give me or show me the statement which tell the communication in TLS is secure? 2. Would anybody mind give me or show me how the call flow in TLS support? (I mean the flow of encrypt call by using TLS) 3. In voIP communication, from where the sniffer can eavesdrop the communication untill TLS is built to protect communication? and by using TLS, is this hole closed ?(so the communication becomes secure?) Those are the questions. I do hope anybody can give me an answer so I can show it to my lecturer to proof that the TLS communication is secure. Please give me a suggestion. Please.. Thank you very much :) Regards, Ferianto --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

3 3

[Serusers] ser new_connection: Access denied for user
by vijay tiwari 21 Oct '06

21 Oct '06

hi problem command ser [root@bishthost sbin]# ser 0(3448) WARNING: fix_socket_list: could not rev. resolve 192.168.5.93 0(3448) WARNING: fix_socket_list: could not rev. resolve 192.168.5.93 Listening on udp: 192.168.5.93 [192.168.5.93]:5060 tcp: 192.168.5.93 [192.168.5.93]:5060 Aliases: stateless - initializing [root@bishthost sbin]# 0(0) Maxfwd module- initializing 0(0) new_connection(): Access denied for user 'root'@'192.168.5.93' (using pass word: YES) 0(0) db_init(): Could not create a connection 0(0) ERROR: uridb_db_ver: unable to open database connection 0(0) ERROR: uri_db:mod_init(): Error while querying table version 0(0) init_mod(): Error while initializing module uri_db ERROR: error while initializing modules when i am login to mysql by this command it working mysql -u root -p password : ************ mysql > mysql > please help in this i am sending my ser.cfg if any change in ser.cfg please let me know thanks vijay --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

1 0

[Serusers] voice_silo table missing. Please help!!
by Meidiana Ten 21 Oct '06

21 Oct '06

Hi there, I've read your post in mailing list about voice_silo table. I setup my SER 0.9.7 and I am trying to activate msilo module to store voice message for offline user. I've checked the SER database and found that there is no table named voice_silo. Please help me with the table description so I can create the table. And please tell me how voicemail is store? ( I mean if it is store directly into voice_silo table using x-lite or we must use other application to input voice then store it into any folder). I really need detail because I'm newbie in Linux. Thanx Meidiana --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 2

[Serusers] Unable to include prolog.html and separator.html
by lin9k＠yahoo.com 20 Oct '06

20 Oct '06

Hi, When I conntect to the serweb the web page shows the warning as below. I am certain the two files prolog.html and separator are in the correct directory and the permission of them and parent directories are correct. Any hints are welcome. Thanks in advance. Warning: virtual() [function.virtual]: Unable to include '/var/serweb/domains/_default/prolog.html' - request execution failed in /var/serweb/html/page.php on line 79 192.168.0.200 user management Warning: virtual() [function.virtual]: Unable to include '/var/serweb/domains/_default/separator.html' - request execution failed in /var/serweb/html/page.php on line 81 Ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] serweb log in failed: Can't parse username and domain part
by lin9k＠yahoo.com 20 Oct '06

20 Oct '06

Hi, Any body know why I get the error message from /var/log/serweb? serweb [info] User login: authentication failed: unsuported format of username. Can't parse username and domain part. I'm using: mysql 3.23.54 serweb-0.9.4 ser 9.9.6 httpd-2.2.3 php-5.1.6 Thanks in advance, Ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Routing script
by Girish Nayak 20 Oct '06

20 Oct '06

I need to rewrite the URI based on the @from field and then route the call to another proxy. How to do it with routing scripts? Is there any documentation available on the routing scripts? Girish

1 0

Re: [Users] Difference between Alias and Listen
by Mark Kent 20 Oct '06

20 Oct '06

Continuing on: >> Basically what I am trying to achieve is to just have the server >> listening on a specific (fixed) IP and just point multiple domain >> names at it's public IP without having to edit the config or >> restart openser. Do you know if this is possible? I still think it is bad practice to list actual domain names in a listen statement. You should list IP addresses. I'ld like to see someone with more experience running openser agree with me. Also I think most of my queries from earlier in this thread remain unanswered, even though Juha took a stab at it he said he couldn't speak authoritatively about some of it. This part here is still unclear to me: == > For example, if you are using the "domain" module then would we == > expect to never have to use "uri==myself" and not have to list aliases? == > (using is_uri_host_local() instead) == == i need to use alias for checking pre-loaded route header containing the == domain name of the proxy. I think that answer translates to "No. You need uri==myself in addition to is_uri_host_local(). The former gets messages directed AT you, the latter gets messages directed at the DOMAINS you support." If so, I'ld like to see an unambigous "Yes, that is correct" from Juha or Bogdan or Klaus... Thanks, -mark

2 1

Re: [Users] Difference between Alias and Listen
by Mark Kent 20 Oct '06

20 Oct '06

>> Basically what I am trying to achieve is to just have the server >> listening on a specific (fixed) IP and just point multiple domain >> names at it's public IP without having to edit the config or >> restart openser. Do you know if this is possible? I'ld like to see someone authoritative say Yes, while waiting for that.... I'll say Yes, using the domain module. I have one listen statement, listing one ip address and one port. I have one alias statement listing the actual hostname of the machine. I have a few domains listed in my db, and I use this code: # checks if the URI either points at us or if it is from a trusted # source. If it does point AT us, then we just return with a positive # value and the caller should continue # route[37] { if (is_uri_host_local()) { return(1); } if (is_from_local()) { # it wasn't destined *for* us, but it's from # someone we trust, so relay it log(1, "from local...\n"); route(21); } else if (allow_trusted()) { # it wasn't destined *for* us, but it's from # someone we trust, so relay it log(1, "from trusted...\n"); route(21); } else { sl_send_reply("403", "Forbidden. We don't relay everything..."); } return(-1); } and I use that at near the top of route[0] route { xlog("L_ERR", "$rm $ru $si $fu\n"); # sanity checks if (!route(39)) { exit; } if (!is_method("REGISTER")) { record_route(); } # loose-route if (!route(38)) { exit; } # meant for us? if (!route(37)) { exit; } [snip: handle different methods] } I would like to see someone authoritative comment on the above. I've got more to say, but I'll put it in another email so it doesn't get lost. -mark

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[Serusers] problem starting in ser
by vijay tiwari 20 Oct '06

20 Oct '06

please help in this commmand : serctl start [root@bishthost ser]# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed command : ser [root@bishthost ser]# ser 0(3666) WARNING: fix_socket_list: could not rev. resolve 192.168.5.93 0(3666) WARNING: fix_socket_list: could not rev. resolve 192.168.5.93 Listening on udp: 192.168.5.93 [192.168.5.93]:5060 tcp: 192.168.5.93 [192.168.5.93]:5060 Aliases: stateless - initializing [root@bishthost ser]# 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: tcp_init: bind(5, 0x8117144, 16) on 192.168.5.93: Address already in use thanks vijay --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Users] Difference between Alias and Listen
by Max Gregorian 20 Oct '06

20 Oct '06

Hi there, Could someone please explain to me the actual difference between *Alias=*and *Listen=*. I have read the documentation ( http://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#alias and http://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#listen) but still cannot distinguish between what the two functions are doing differently. Do I *need* to use both Alias and Listen at the same time in the config? For example, if I have an openser server responsible for multiple domains * sip1.foobar.com*, *sip2.foobar.com*, etc and also listening in on multiple ports (*5060*, *5061*, *5062*), would I have to add the following to my openser.cfg? listen=udp:localhost:5060 /* Loopback */ listen=udp:localhost:5061 listen=udp:localhost:5062 listen=udp:192.168.1.101:5060 /* LAN IP of the server */ listen=udp:192.168.1.101:5061 listen=udp:192.168.1.101:5062 listen=udp:sip1.foobar.com:5060 /* Domain 1 */ listen=tcp:sip1.foobar.com:5060 listen=udp:sip1.foobar.com:5061 listen=tcp:sip1.foobar.com:5061 listen=udp:sip1.foobar.com:5062 listen=tcp:sip1.foobar.com:5062 listen=udp:sip2.foobar.com:5060 /* Domain 2 */ listen=tcp:sip2.foobar.com:5060 listen=udp:sip2.foobar.com:5061 listen=tcp:sip2.foobar.com:5061 listen=udp:sip2.foobar.com:5062 listen=tcp:sip2.foobar.com:5062 alias="sip1.foobar.com:5060" /* Aliases for Domain 1 */ alias="sip1.foobar.com:5061" alias="sip1.foobar.com:5062" alias="sip2.foobar.com:5060" /* Aliases for Domain 2 */ alias="sip2.foobar.com:5061" alias="sip2.foobar.com:5062" alias="111.222.333.444>" /* IP OF SERVER */ Now if I understand correctly then *if (uri==myself) { }* should now be true for each of these domains.

2 2

[Users] new SEMS
by Richard Bennett 20 Oct '06

20 Oct '06

Hi, Has anyone tried the new SEMS RC1 with Openser yet? Is there any chance this might work? see: http://www.iptel.org/howto/sems_application_development_tutorial Thanks, Richard.

2 1

[Users] Generating an INVITE from Openser's dialplan
by Antoine Fressancourt 20 Oct '06

20 Oct '06

Hello, I plan to set up a dialplan where an invite is generated to an application server when a user agent emits a given DTMF, sent to OpenSER as a SIP INFO message. This way, I want to do 3rd party call control. To begin, I have a rather simple question : how to generate an INVITE message from OpenSER's dialplan? Is there a particular module? Should I use a dedicated application server acting as a B2BUA receiving the INFO messages and generating the INVITEs in return ? Thank you in advance for your help, Antoine

3 2

RE: [Serusers] problem in installing Mysql
by vijay tiwari 20 Oct '06

20 Oct '06

hi all thanks alex <kingofthemods(a)gmail.com> wrote: v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} Looks like mysql isent started. Simple error start it through init.d or restart I forget which --------------------------------- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of vijay tiwari Sent: 20 October 2006 12:27 To: serusers(a)lists.iptel.org Subject: [Serusers] problem in installing Mysql hi all i have change makefile and reinstall the binaries(make install) and after this command [root@bishthost ser-0.9.6]# make dbinstall Initializing ser database scripts/ser_mysql.sh create MySql password for root: # given the password Domain (realm) for the default user 'admin': 192.168.5.93 # given the domain 192.168.5.93 creating database ser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) make: *** [dbinstall] Error 1 # after that getting this error please help in this thanks vijay --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

1 0

Re: [Users] detecting BYE
by Ramin Dousti 20 Oct '06

20 Oct '06

OK, thanks, Nicolas. I just confirmed the ACKs can be detected the same way. Now another question: is there anyway to generate an ACK locally as an answer to a BYE? I checked the SL module but it only does reply and error but it doesn't craft a message.on its own like "ACK". Ramin On 10/19/06, Nicolas Palix <nicolas.palix(a)labri.fr> wrote: > It should also work. > > Have you check with tcpdump or wireshark that the ACK pass through your > OpenSER server ? > > PS: May you keep the discussion on the mailing list... > > > Le jeudi 19 octobre 2006 16:28, vous avez écrit: > > OK, I'm able to see the BYE's. > > How can I intercept the ACK's? I tried the same way but I don't see them? > > > > On 10/19/06, Nicolas Palix <nicolas.palix(a)labri.fr> wrote: > > > Hi, > > > > > > You may use the Record Route and then check the BYE method. > > > > > > Le mercredi 18 octobre 2006 22:21, Ramin Dousti a écrit: > > > > Hi, > > > > > > > > Is there any way to detect a BYE while t_relay() has been called for > > > > the corresponding INVITE? In what block? > > > > > > > > TIA, > > > > Ramin > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users(a)openser.org > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > > > Nicolas Palix > > > http://phoenix.labri.fr/people/palix/ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > -- > Nicolas Palix > http://phoenix.labri.fr/people/palix/ > -- Ramin

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[Serusers] problem in installing Mysql
by vijay tiwari 20 Oct '06

20 Oct '06

hi all i have change makefile and reinstall the binaries(make install) and after this command [root@bishthost ser-0.9.6]# make dbinstall Initializing ser database scripts/ser_mysql.sh create MySql password for root: # given the password Domain (realm) for the default user 'admin': 192.168.5.93 # given the domain 192.168.5.93 creating database ser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) make: *** [dbinstall] Error 1 # after that getting this error please help in this thanks vijay --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

3 2

[Serusers] New SER release 0.9.6 doesn't work
by paolo acquaviva 20 Oct '06

20 Oct '06

Hi, I would upgrade SER to 0.9.6 release. I follow all instructions also I make install library like mysql, auth_radius and so. After the installation I try to start with a new release there isn't error message in /var/log/messages but the SER doesn't start. Why? Can anyone say me anything? Thanks. Regards, Paolo

2 1

[Serusers] ser sgml documentation
by Mark Price 20 Oct '06

20 Oct '06

What is the best way to view the sgml documentation included in the ser-0.9.6 tarball? Preferrably a website or such where it is already rendered? Thanks Mark Price

2 1

[serusers] serctl ping response - 477
by Kamal.Mann＠t-systems.com 20 Oct '06

20 Oct '06

Hi All I am trying to implement blind call-forwarding. But from ser logs m getting "Oct 19 14:48:52 localhost /usr/sbin/ser[3971]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net". And when I am trying 'serctl ping sip:sony@imsfw.de' its giving me 477!!! Why it is so?? Even when I enable listen=IP_address into my ser.cfg my clients unable to register with ser but if I comment it out then it works fine & starts listening on my IP. Please give me your valuable suggestions. Thanks in anticipation Kamal Mann

2 2

[Serusers] issue w/ permissions module
by Corey 19 Oct '06

19 Oct '06

Hello! I'm using postgresql for the db backend for my ser instance, after creating and populating a few obvious tables by handI seem to have it "mostly working". ( using ser-0.9.4 ) However, I'm stuck on the following: Oct 18 21:09:36 server /usr/sbin/ser[23595]: ERROR: permissions: trusted_reload(): Database problem Oct 18 21:09:36 server /usr/sbin/ser[23595]: init_trusted(): Reload of trusted table failed Oct 18 21:09:36 server /usr/sbin/ser[23595]: Error while initializing allow_trusted function Scratching my head, trying to figure out what might be causing that, but with no success - any clues? What I imagine is the relevant info: server ser # psql -U ser -W ser Password for user ser: Welcome to psql 8.1.4, the PostgreSQL interactive terminal. ser=# \dt List of relations Schema | Name | Type | Owner --------+------------+-------+------- public | subscriber | table | ser public | trusted | table | ser public | version | table | ser (3 rows) ser=# \d trusted Table "public.trusted" Column | Type | Modifiers --------------+-----------------------+----------- src_ip | character varying(39) | not null proto | character varying(4) | not null from_pattern | character varying(64) | not null Indexes: "trusted_pkey" PRIMARY KEY, btree (src_ip, proto, from_pattern) Am I missing a table, or a column somewhere? Is this a ser issue, or a db/sys-admin/permissions issue? Should I also provide my ser.cfg? Many thanks for any help/advice! ( also, is it possible to use the ser-0.10.x ser_postgres.sh/ser_dbtext.sh scripts for ser-0.9.4, rather than doing it all by hand? )

4 4

[Users] flags and pseudo variables problem
by Fatih Cerit 19 Oct '06

19 Oct '06

Hi all When the save function called the value of FLAG is 1 and when we want to get this from pseudo $mf the value is 64. I only set flag to 6. Is there any conversation at here or something wrong?? Thanks Fatih Cerit

2 1

[Users] Problem with Call Forwarding
by Darryl Ross 19 Oct '06

19 Oct '06

Hey All, I have tried to set up forwarding using a document for SER which I found at http://siprouter.onsip.org/doc/gettingstarted/ch10.html. In trying to get it to work though I am having a couple of issues: 1) Call forward immediate and not available works, but the billing party is set to the caller and it needs to be rewritten to be the account doing the forward. 2) Call forward busy and no answer don't work at all. The caller's phone drops the call and the forwarding destination phone does not ring at all. I believe the solution to both of those problems is the same: find out how branching works and set it up properly, but I am struggling to find some documentation to help. Does anyone have a sample config they can send me. I'm sure it would help enormously! I'm using OpenSER 1.1.0 with mediaproxy 1.7.2 on debian sarge. Regards Darryl

2 1

[Users] Mediaproxy newbie
by Marnus van Niekerk 19 Oct '06

19 Oct '06

Hi, I am trying to set up openser with mediaproxy (at xx.xx.xx.133) to route calls from UA behind NAT to asterisk as voicemail (at xx.xx.xx.134) and PSTN gateways (at xx.xx.xx.32) I can see in the SDP payload that the RTP is being sent from asterisk to mediaproxy, but in sessions.py it shows the private ip not the public one and I have one way audio. Can anybody help please. opnser.cfg below. Marnus -- debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL6 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" # Logging loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) #tm timeout for voicemail params modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 35) modparam("tm", "noisy_ctimer", 1) # parms for NAT/mediaproxy modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/openser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/openser/rtp-clients") modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route { #check for old messages: could mean a problem with the DNS entries or some other loop-causer... if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", "WARNING: Too many hops\n"); sl_send_reply("483", "Too many hops, forward count exceeded limit\n"); return; }; #check for extremely large messages; we don't need a sip dos attack if (msg:len >= 2048) { xlog("L_WARN", "WARNING: Message too large, &>= 2048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; }; # Track what is happening xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n"); #record everything besides registers and acks if(method!="REGISTER" && method!="ACK") { setflag(1); }; # Record Route Section if (method=="INVITE" && client_nat_test("3")) { record_route_preset("xx.xx.xx.133:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); } # Call tear down section if (method=="BYE" || method=="CANCEL") { end_media_session(); } #do not send to voicemail if BYE or CANCEL #is used to end call before user pickup or timeout if(method=="CANCEL" || method=="BYE") { setflag(10); }; #grant route if route headers already present if (loose_route()) { # May need client_nat_test & use_media_proxy here... route(1); return; }; #Always require authentication, which could result in a PSTN if (method=="REGISTER") { if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if(!www_authorize("domain.tld", "subscriber")) { www_challenge("domain.tld", "0"); return; } else { if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; #Save into user database, used below when checking if user is available xlog("L_INFO", "REGISTER: User $fu Authenticated Correctly\n"); save("location"); return; }; }; if (method=="INVITE") { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(uri=~"sip:\*86@.*") { #authorize if a call is going to VM if(!proxy_authorize("domain.tld", "subscriber")) { proxy_challenge("domain.tld", "0"); return; }; xlog("L_INFO", "CALL: Call from $fu to check voicemail\n"); rewritehostport("vm.domain.tld:5060"); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_ERROR", "ERROR: User $tu Not Found\n"); return; }; } else { #authorize if a call is going to PSTN if(!proxy_authorize("domain.tld", "subscriber")) { proxy_challenge("domain.tld", "0"); return; }; #Call destination is PSTN, so send it to the gateway xlog("L_INFO", "CALL: PSTN $tu from $fu \n"); rewritehostport("ast1.domain.tld:5060"); }; }; #Make sure that all subsequent requests go through us; #done at the top already #record_route(); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_ERROR", "ERROR: User $tu Not Found\n"); return; }; } else { #Call destination is PSTN, so send it to the gateway xlog("L_INFO", "CALL: PSTN $tu from $fu \n"); rewritehostport("ast1.domain.tld:5060"); }; #record_route(); }; #ALL PROCESSING IS DONE, SO ROUTE route(4); route(1); } route[1] { #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); # voicemail if failure }; if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } failure_route[2] { if(!t_was_cancelled() && !t_check_status("407")) { revert_uri(); rewritehostport("vm.domain.tld:5060"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures.

2 1

Re: [Users] OpenSER not working after running for several days
by Mark Kent 19 Oct '06

19 Oct '06

>> Locking it to 10/full was what the MCI technician recommended after >> I opened a ticket because the line was dropping to 500Kbs >> up/down. This was a 4Mbit up/down line burstable to 10. They said >> it was not possible to increase the capacity of this line, so I >> guess they are using their old 10Mbit equipment for this. Since >> locking it like they asked speed has been fine, OK, it wasn't clear that you were talking about a specialized circuit delivered from a telco. BTW, this phrase "the line was dropping to 500Kbs up/down" implies that there was some protocol synching taking place that negotiated the speed at that level (like a dsl line). It would be more accurate to type "my observed throughput was only 500Kbs up/down." >> and the cable is a 2 meter factory made one plugging into the >> datacenter patch-panel, so I don't think that's a problem... My reference to the cable was for a situation where we're talking about a real local area network, where you might have a central switching fabric and machines distributed perhaps throughout a building and was meant to apply to the cable run from the main switches to the openser box. So, now I don't get your set-up... when you typed "I had to lock-down the network card to 10mbit full-duplex" and then the bit above about the MCI technician and the 4Mbit circuit etc. then this sounds like you have a box running openser that is directly plugged into a metro-lan-style connection that is hardcoded at the provider end to 10/full. And yet, when you type "datacenter patch-panel" this implies that there is a local area network which implies some sort of central switching fabric and then when I consider "old 10Mbit equipment" together with the phrase "datacenter" my jaw hits the ground... Where is your server? Thanks, -mark

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[Users] Difference between Alias and Listen
by Mark Kent 19 Oct '06

19 Oct '06

>> Could someone please explain to me the actual difference between *Alias=*and >> *Listen=*. I have read the documentation ( >> [snip] >> listen=udp:sip1.foobar.com:5060 /* Domain 1 */ Ahh, this links back to your previous message and adds a little light. IMO, it is not good practice to put domain names for listen. I would only list IP addresses, and that list should be relatively small because you don't need the IP address to distinguish between clients. For example, I just have one listen command in my config. For a critical service like this, either hard-coding in the IP addresses to listen on, or be willing to listen on all attached interfaces, should be Standard Operating Procedure (I vote for the former). >> alias="sip1.foobar.com:5060" /* Aliases for Domain 1 */ Don't put the port in there. Just list the domains that the server is responsible for. >> Now if I understand correctly then *if (uri==myself) { }* should >> now be true for each of these domains. I think this should be explained more fully by someone in the know... For example, if you are using the "domain" module then would we expect to never have to use "uri==myself" and not have to list aliases? (using is_uri_host_local() instead) An earlier hint from someone here to me said something like "openser doesn't know that the message is directed at itself" and I wonder how the use of the "domain" module affects things *inside* the openser core. And then there is the environment variable SIP_DOMAIN... does that have any bearing on the function of openser or is it just used for the supporting scripts (openserctl). What is the role of this variable in a multi-domain setting? -mark

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[Serusers] Serweb: Forbidden, You don't have permission to access /serweb/admin/index.php
by lin9k＠yahoo.com 19 Oct '06

19 Oct '06

Hi, After followed the INSTALL file set up the serweb. Have configure Alias /serweb "/var/serweb/html" in httpd.conf But while trying to browse http://192.168.0.200/serweb/admin/index.php the error message showed up on the broswer: Forbidden You don't have permission to access /serweb/admin/index.php on this server If I just browse form the local machine as: /var/serweb/html/admin/index.php the content of index.php would show up What could be wrong of my setting? I have: httpd-2.2.3 php-5.1.6 serweb-0.9.4 installed in my redhat 9.0 machine where ser-0.9.6 has been working fine. Thanks a lot for help. Ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

Re: [Users] Difference between Alias and Listen
by Mark Kent 19 Oct '06

19 Oct '06

>> Thus, openser internally has a list of aliases it is responsible of. >> This list includes: >> [snip] >> - all the domain names found out by reverse lookups of the listening IPs I see that Richard Bennett had discovered that as he earlier typed: == it seems something was trying to look-up the aliases == that that server itself was listening on where he likely meant IP addresses and not aliases. I have to say that this is a big mistake. Doing the lookups is a bad idea and basing any sort of operational aspects of openser on the domains culled from such reverse lookups is even worse. The list of supported domains should be purely under the explicit configuration control of the person running the openser box... either with explicit "alias" statements, or with the use of the "domain" module. I could support this claim with a lengthy, experience supported, argument. But I'll save time with just this: 70.22.4.4.in-addr.arpa name = wbar1.sjo1-4-4-022-070.sjo1.dsl-verizon.net. -mark

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Re: [Users] OpenSER not working after running for several days
by Mark Kent 19 Oct '06

19 Oct '06

The two messages I read with the above subject seem to overlook some important concepts: a) Like many many servers, openser is a high-level application. It depends on a complicated under-world of services and protocols. The operation of it depends on many external elements, all of which should be in good working order. b) DNS resource records (RR) have time-to-live (TTL) attributes. Any application may cache RR up to the length of time specified by the TTL. When you restart something, you typically blow away the cache and start over. In general, most applications do not cache general name lookups, since that is a lower-level function. However, an application caching the hostname it's on and/or IP addresses to bind to is standard practice. >> I have come to realise that it seems as though once loaded into memory >> and running, openser "caches" the config file and also does not appear >> to re-check for changes in network conditions Yes, isn't that great! It shouldn't check until either told to by a human or when the underlying protocol indicates that it is time for a refresh (like a DNS TTL expiring). Regarding this: > I had to lock-down the network card to 10mbit full-duplex. It was > autoselecting 100mbit half-duplex. > Hope that helps someone... sometime... Autoselecting 100/half probably means that the link auto-negotiation failed, as that is the default. It might fail because the other side was nailed down, perhaps to 10/full. That seems like an unlikely choice... but one reason why it might be nailed at 10mbits is because someone before you couldn't get it to run well at 100mbits and discovered that setting it to 10 was a "solution." If that was the case, I'ld guess that the cable between the switch and the server is lengthy and that it was not tied down in the prescribed order (wO, O, wG, Bl, wBl, G, wBr, Br) [or swap O and G]. I apologize if this email comes across as a little snooty. This is my first email of the day and I often have trouble phrasing things in a nice way prior to my coffee kicking in. -mark

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RE: FW: [Users] OpenSER config help to talk to PSTN..
by Nick De Cristofaro 19 Oct '06

19 Oct '06

I know, but I want it like this. It is supposed to talk to a remote mysql server. As configured in openserctlrc file. But after doing some fiddling around I noticed this error actually occurs because of a misconfiguration in the config file itself because if I use the default openser config file it works fine and uses the remote sql server, there isn't anything I can find in my other config file that would tell it to use the local one instead of a remote one though. Nick -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: October 19, 2006 5:08 PM To: Nick De Cristofaro Cc: users(a)openser.org Subject: Re: FW: [Users] OpenSER config help to talk to PSTN.. maybe there is no local mysql server on your machine...... regards, bogdan Nick De Cristofaro wrote: > Found the issue now, it was an original problem I had but I'm not sure > how it went away in the first place. There is a problem with the mysql > connection, which is located on another server. > > If I attempt to just run the mysql client manually it still does not > work.. > > ERROR 2002 (HY000): Can't connect to local MySQL server through socket > '/tmp/mysql.sock' (2) > > seen this error a million times on google but couldn't really find any > fixes that worked for me? This error occured earlier on when I first > configured openser but It went away and I'm not sure how it happened. > > I rebooted the machine and i'm guessing this started again due to it > > Thanks > Nick >

2 1

[Users] Media proxy daemon and sems problem. RTPProxy also concerned ?
by Christophe Irles 19 Oct '06

19 Oct '06

Hi, According the RFC, it's possible to send a INVITE without any SDP. In this case, the ACK contains the SDP after the 200 OK (with SDP) send by the callee. In this situation, the RTP/RTCP is not dealed correclty by the media proxy and sems i.e impossible to have the audio. Do you think this situation is possible ? The RFC can expose some case never used for example. Do you know if the RTPProxy daemon is also concerned by this problem ? Thanks, Christophe

2 1

FW: [Users] OpenSER config help to talk to PSTN..
by Nick De Cristofaro 19 Oct '06

19 Oct '06

below is my config file i modified it a bit.. still pretty basic. trying to get basic connectivity to a PSTN gateway. I modified an example file. errors I get are: bash-3.00# openser -c 0(6841) parse error (98,21-28): parse error 0(6841) parse error (98,29-30): bad arguments 0(6841) parse error (121,33-36): parse error 0(6841) parse error (121,51-52): bad forwardargument 0(6841) parse error (139,55-56): unknown command, missing loadmodule? 0(6841) parse error (145,55-56): unknown command, missing loadmodule? 0(6841) parse error (151,55-56): unknown command, missing loadmodule? 0(6841) parse error (163,10-11): parse error 0(6841) parse error (163,10-11): ERROR: bad config file (9 errors) # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) alias=ip /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ------------------ logs ------------------------------------------- log_facility=LOG_LOCAL7 #Add the xlog module: loadmodule "/usr/local/lib/openser/modules/xlog.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) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); return; }; /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); return; }; /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip:\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { forward(uri:host, uri:port); }; return; }; # account completed transactions via syslog setflag(1); # authorize only for INVITEs -- RR/Contact may result in weird # things showing up in d-uri that would return our logic; our # major concern is INVITE which causes PSTN costs if (method=="INVITE") { # does the authenticated user have a permission for local # calls (destinations beginning with a single zero)? # (i.e., is he in the "local" group?) if (uri=~"sip:0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "No permission for local calls"); return; }; # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permission for LD "); return; }; # the same for international calls (three zeros) } else if (uri=~"sip:000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International permissions needed"); return; }; # everything else ( e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); return; }; }; # INVITE to authorized PSTN }; # authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("ip:5060"); # forward the request now if (!t_relay()) { sl_reply_error(); return; }; } bash-3.00#

2 5

[Users] "invalid cseq for aor"
by Natambu Obleton 19 Oct '06

19 Oct '06

I read in the recent archives and I say some talk on this, but then I found this. [Devel] [ openser-Bugs-1493632 ] ERROR:registry:update_contacts: invalid cseq for aor http://lnk4.us/PeLQ I got the newest version out of CVS and my polycomm phone still has this problem. 0(25130) ERROR:registrar:update_contacts: invalid cseq for aor <2012033002> Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

2 3

Re: [Serusers] rfc 3262 support
by Jiri Kuthan 19 Oct '06

19 Oct '06

At 07:40 19/10/2006, Kamal.Mann(a)t-systems.com wrote: >Hi All >Do SER supports 'rfc 3262 Reliability of Provisional Responses in SIP'? Of course. (SER supports all end-device features it doesn't care of, including INFO, video, gaming, colorful displays, etc.) >Is there any list of standards supported by SER available?? RFC3261, 3263,3265,3325,RFC4028,3581,3428,3903. some of those are implemented as a scripting feature. + possibly parts of some additiional ones. -jiri

1 0

[Users] Via: ...;received=...;rport=... processing
by Jeremie Le Hen 19 Oct '06

19 Oct '06

Hi list, I checked the OpenSER core cookbook, but I am unsure how to handle the "received" and "rport" parameters in the "Via" header. Is it used automatically ? For now, I am using the registrar module to save the location, but I think I can get rid of it and simply use the above parameters. Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 1

[Users] OpenSER not working after running for several days
by Papadopoulos Georgios 19 Oct '06

19 Oct '06

Hello, We have been using OpenSER for almost a year in a production system. So far the only problem is that sometimes it stops processing the requests. It does not happen very frequently, maybe every 10-15 days, sometimes it is more, sometimes it is less. When it happens, the process is still running, I can see it when I do "ps". Also netstat shows that it is listening on port 5060. So everything looks normal, except that OpenSER does not seem to process the requests. I use ngrep and I see that there isn't a single packet being sent out. At that point the only thing that cures the problem is to kill the process and start it again. A couple of weeks ago, in an effort to solve the problem I upgraded from 1.0 to 1.1. It ran fine for several days but eventually the same thing happened. I have no idea what to check for. Has anyone else seen this problem? Is there anything I can do to figure this out? The only solution that I can think of is to have OpenSER restarted every 24 hour or so. But I really do not want to end up there. Thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Users] How to configure acc modue
by Madhusudan.Jarhad＠lntinfotech.com 19 Oct '06

19 Oct '06

Dear All, Here is very basic question from me. I have installed openser 1.1.0. I want to record the CDR / Accounts of calles and messages being processed. I have acc module in my opense.cfg and parameter setting is as follows # ----------------- SETTING MODULE-SPECIFIC PARAMETERS --------------- ## - - - - - - - - - - XLOG PARAMETERS - - - - - - - - - - - - - - - modparam("xlog", "force_color", 0) # - - - - - - - - - USRLOC PARAMS - - - - - - - - - - - - - - - - - - #modparam("usrloc", "db_mode", 1) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # - - - - - - 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") #modparam("alias_db", "user_column", "username") modparam("auth_db", "password_column", "password") ## - - - - - AUTH RADIUS - - - - - #modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("auth_radius", "service_type", 15) ## - - - - - RR MODULE PARAMETERS - - - - - # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 1) ## - - - - - ACCOUNTING PARAMETERS - - - - - - - - modparam("acc", "detect_direction", 1) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels",1) modparam("acc", "report_ack", 1) modparam("acc", "early_media", 1) modparam("acc", "detect_direction", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) #modparam("acc", "log_extra", "ua=$hdr(User-Agent);uuid=$avp(i:123)") modparam("acc", "service_type", 15) # user 16 to set to Radius modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_missed_flag", 3) modparam("acc", "acc_sip_method_column", "sip_method") modparam("acc", "acc_from_uri_column", "from_uri") modparam("acc", "acc_fromtag_column", "fromtag") modparam("acc", "acc_to_uri_column", "to_uri") modparam("acc", "acc_totag_column", "totag") #modparam("acc", "acc_cseqno_column", "cseq_no") modparam("acc", "acc_sip_callid_column", "sip_callid") modparam("acc", "acc_time_column", "time") # for mysql-based accouting, use the names of the columns # for RADIUS-based accouting, use the names of the RADIUS AVPs #modparam("acc", "multi_leg_info","RAD_LEG_SRC=$avp(src);RAD_LEG_SRC=$avp(dst)") #modparam("acc", "multi_leg_info","leg_src=$avp(src);leg_dst=$avp(dst)") #modparam("avpops", "avp_aliases", "uuid=I:50") modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_config", "/etc/openser/radius/client.conf") #modparam("acc", "radius_flag", 2) #modparam("acc", "diameter_missed_flag", 3) #modparam("acc", "radius_extra", "Sip-RPid=$avp(s:rpid) Source-IP=$si; Source-Port=$sp" # Canonical-URI=$avp($can_uri); \ # Billing-Party=$avp($billing_party); \ # Divert-Reason=$avp(s:divert_reason); \ # X-RTP-Stat=$avp(s:rtp_statistics); \ # From-Header=$hdr(from); \ # User-Agent=$hdr(user-agent); \ # Contact=$hdr(contact); \ # Event=$hdr(event)") ## - - - - - - JABBER SERVER CONFIGURATION - - - - - - - - - modparam("jabber", "jdomain", "jabber.sipserver.com=*") modparam("jabber", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("jabber", "jdomain", "sipx.vshodc.lntinfotech.com") ## - - - - - - - - - - - SIPTRACE MODULES PARAMETERS - - - - - - - - - modparam("siptrace", "trace_flag", 0) modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "trace_on", 1) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "duplicate_uri", "sip:172.17.85.51:5888") # ------------------------- request routing logic ------------------- # main routing logic route{ ## ADDED BY MADHUSUDAN # avp_write("caller_id", $avp(name); #"$uuid"); # avp_printf("i:20", "$avp($uuid): $fu"); # xdbg("$Cbgi:20$Cxx [$avp(i:20)] $Cbrcseq$Cxx=[$hdr(cseq)]\n"); # setflag(1); /* set for accounting (the same value as in log_flag!)*/ # t_relay(); /* enter stateful mode now */ ## sip_trace(); # xlog("L_ERR", "time [$Tf] method <$rm> r-uri <$ru> 2nd via <$hdr(via[1])>\n"); # setflag(1); /* set for accounting (the same value as in log_flag!)*/ # t_relay(); /* enter stateful mode now */ # if (t_newtran()) { # log("UAS logic"); # t_reply("999","hello"); # } else sl_reply_error(); ################ if (t_local_replied("all")) { log ("no reply received\n"); } # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } onreply_route[1] { if (t_check_status("1[0-9][0-9]")) { setflag(1); log("provisional reply received\n"); if (t_check_status("183")) drop; } if (t_was_cancelled()) { log("transaction was cancelled by UAC\n"); } } ################################################################### My need is to get the CDRs for events and volume like MESSAGES and download/file transfer respectively. Please help ______________________________________________________________________

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[Users] OpenSER Summit: Agenda & Special guests
by Daniel-Constantin Mierla 19 Oct '06

19 Oct '06

Hello everybody, we are now one week and a half since the registration to OpenSER summit opened and two weeks and a half to the start of event. At this moment we have more than 60 registered people and special guests. The numbers promise a successful event, having there people from all categories of VoIP business: developers, service providers, device vendors, investors and standardization bodies. The agenda is now in very goos shape, covering many aspects of VoIP and OpenSER, Please have a loot at: http://www.openser.org/index.php?option=com_content&task=view&id=60&Itemid=… Day by day new special guests join the event. Update list is published at: http://www.openser.org/index.php?option=com_content&task=view&id=61&Itemid=… Within the summit will be launched a bunch of new services, applications and extensions. We will keep you updated with these events in the next days. Because of the logistics and materials we have to prepare, is very likely that the registration will close several days before the event. The ones that decide to come afterwards will not benefit of the event's offerings (meals, brochures, CDs...), but they can attend. To register, fill and submit the form at: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… There is no charge to attend the event. More details on: http://www.openser.org Cheers, Daniel

1 0

[Users] Mediaproxy module and new equipment
by Artur Yakupov 19 Oct '06

19 Oct '06

Hi, all! Preamble: I got new equipment - videophone. I tried to make it work with OpenSER (version 1.1.0). Schema: VIDEOPHONE -> OpenSER -> VIDEOPHONE I tried to make simple call. It works but not behind NAT. With rtpproxy OpenSER logs show such messages: ERROR: force_rtp_proxy2: can't extract media port from the message Calls last only few seconds than it get disconnected. With mediaproxy OpenSER logs show following: Oct 16 18:01:22 sip mediaproxy[20251]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected '' at 0x68 2ecc> (exceptions.ValueError:invalid literal for int(): 5012/1 [/usr/lib/python2.3/asyncore.pyc|readwrite|86] [/usr/lib/python2.3/asyncore .pyc|handle_read_event|390] [/usr/local/mediaproxy/modules/rtphandler.py|handle_read|441] [/usr/local/mediaproxy/modules/rtphandler.py|pro cess|531] [/usr/local/mediaproxy/modules/rtphandler.py|__init__|988] [/usr/local/mediaproxy/modules/rtphandler.py|setCaller|880]) I found that source of these troubles is that videophone generates INVITE`s in different format than for example Sipura does. Media streams are described like these: m=audio 5012/1 RTP/AVP 98 0 8. m=video 5010/1 RTP/SAVPF 99. As you can see, sub-field "port" is not in usual format, but it contains number of ports according to RFC2327: m=<media> <port>/<number of ports> <transport> <fmt list> So, I spended few hours examinating sources of rtpproxy and mediaproxy modules and made small trick (patch) to mediaproxy module. I didn't understand logic of rtpproxy module because lack of time. Here it is: --- ./modules/mediaproxy/mediaproxy.c 2006-03-24 21:04:56 +0300 +++ ./modules/mediaproxy/mediaproxy_new.c 2006-10-17 17:53:54 +0400 @@ -825,7 +825,7 @@ getMediaStreams(str *sdp, str *sessionIP, StreamInfo *streams, int limit) { str tokens[2], block, zone; - char *ptr, *sdpEnd; + char *ptr, *sdpEnd, *tmpptr; int i, count, streamCount, result; sdpEnd = sdp->s + sdp->len; @@ -845,6 +845,9 @@ "line in SDP body\n"); return -1; } + + tmpptr=strchr(tokens[1].s,'/'); + if ((tmpptr-tokens[1].s)<tokens[1].len ) tokens[1].len=tmpptr-tokens[1].s; streams[i].type = tokens[0]; streams[i].port = tokens[1]; It works for me, but I know it is not correct solution. I hope my small investigation would be useful. Thanks for your attention. With best regards, Artur Yakupov

1 0

[Users] detecting BYE
by Ramin Dousti 19 Oct '06

19 Oct '06

Hi, Is there any way to detect a BYE while t_relay() has been called for the corresponding INVITE? In what block? TIA, Ramin

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[Users] Openserctl add: Plase help.....
by Ferianto siregar 19 Oct '06

19 Oct '06

Dear all, Thank you very much for this chance. Thank you very much for time. All, I have a problem in understanding the output when running openserctl add command (for adding new user). After, I add new user (success), Why this message always appear? What does it mean? "0\n". "1.00\n". "0\n". "128\n". "4294967295\n"; Because I see that this number never changed. I always get this number when success to add new user. I do hope anybody can give me information, what do those numbers mean? Thanks Regards, FERIANTO --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

2 2

[Serusers] Mediaproxy port question.
by Ricardo Martinez 18 Oct '06

18 Oct '06

Hello Iptel users. I'm having a problem with my mediaproxy box. I have a NAT'd client (a softphone) calling to a SIP-H323 conversor through my SER-Mediaproxy box and then to a Cisco Call Manager (it's a very weird configuration). Anyway, the problem presents when the 200 - OK message arrives from my conversor, the SDP part indicate the media address (10.10.148.230) and port (16952) where the remote endpoint will listen RTP, despite of this the mediaproxy open a session for the port 17360. Please check the log from the mediaproxy and the execution of the "sessions.py" utility, plus the OK message. U 100.100.154.36:5060 -> 100.100.148.246:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.148.246;branch=z9hG4bK67ea.aa12dc97.0. Via: SIP/2.0/UDP 192.168.1.101;received=100.100.148.248;rport=46096;branch=z9hG4bKc0a801650000001d4533a0ee00007c2900000011. Record-Route: <sip:100.100.154.36;ftag=2214787970;lr=on>. Record-Route: <sip:7072408196@100.100.148.246:5060;nat=yes;ftag=2214787970;lr=on>. From: "unknown"<sip:5502408196@sip.desa.mydomain.net>;tag=2214787970. To: <sip:7072408196@sip.desa.mydomain.net>;tag=2260729709. Call-ID: A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101. Cseq: 2 INVITE. Date: Mon, 16 Oct 2006 15:10:37 GMT. Server: siph323. Content-Type: application/sdp. Content-Length: 162. Contact: sip:7072408196@100.100.154.119. . v=0. o=100.100.154.119 1 505533324 IN IP4 100.100.154.119. s=SIP Library call. c=IN IP4 100.100.148.230. t=3370000237 0. m=audio 16952 RTP/AVP 0. a=rtpmap:0 PCMU/8000. Oct 16 12:12:00 sip mediaproxy[18289]: lookup A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101 100.100.148.230:16952:audio 100.100.154.36 sip.desa.mydomain.net local sip.desa.mydomain.net unknown siph323 info=from:5502408196@sip.desa.mydomain.net,to:7072408196@sip.desa.mydomain.net,fromtag:2214787970,totag:2260729709 Oct 16 12:12:00 sip mediaproxy[18289]: execution time: 0.48 ms Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: caller signed in from 100.100.148.248:49154 (RTP) (will return to 100.100.148.248:49154) Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: called signed in from 100.100.148.230:17360 (RTP) (will return to 100.100.148.230:17360) [root@sip ser]# /usr/src/mediaproxy-1.7.2/mediaproxy/sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------------------ 100.100.148.248:49154 - 100.100.148.246:60022 - 100.100.148.230:17360 active 0'39" G711u Audio 336.91k/669.53k/332.62k Total traffic: 79.69kbps/78.12kbps/157.81kbps (in1/in2/out) Session count: 1 Proxy version: 1.7.2 Doing more debugs i discover the RTP traffic coming from the 100.100.148.230 it has source port 17360, but the listening port is not the same (i guess is the one indicated in the SDP). Why mediaproxy assumes the 17360 port is for listening too and ommit the SDP 16952 port ? Is this maybe an asymmetric regarding the RTP media?. If so, i added this line into the rtp-asymmetric-clients file: # # Lines starting with a '#' character or empty lines are ignored. # Put a User-Agent name or regular expression per line. # Check is case insensitive. # # This file should only list SIP clients that are asymmetric regarding # the RTP media streams. For clients that are asymmetric regarding the SIP # signaling, use the equivalent SIP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # siph323 But not solve the problem. Can someone help me here? Thanks in advance Regards Ricardo Martinez.- RedVoiss.

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[Serusers] Needs help in integrating ser with radius
by Lokesh Kumar 18 Oct '06

18 Oct '06

Hello List, Can any one please suggest , how to use AAA in ser with the help of free radius. I had included the radius module in the ser.cfg file, I would like to know what other files needed in the configuration. Is there any How to available to do that Thanks Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.4/480 - Release Date: 10/17/2006

1 0

[Users] Pre-paid?
by moacir FERRARI NETO 18 Oct '06

18 Oct '06

Hi ! Which software I can use for service of I pre-paid with openser? Please help me ! Thanks. Moacir. _______________________________________________________ O Yahoo! está de cara nova. Venha conferir! http://br.yahoo.com

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Re: [Users] calls being disconnected
by Richard Bennett 18 Oct '06

18 Oct '06

On Wednesday 18 October 2006 05:03, senthil vinayagam wrote: > Hi Richard, > > If you are using openser with TCP/TLS enabled then Openser will close the > tcp/tls connection after couple of minutes of inactivity by the phone. > Either we have to change the code of openser to increase the tcp > inactivity timer or change phone to re-register every 80 seconds or so. > Otherwise the tcp/tls connection shall get closed and no further calls can > be placed after that. > > My suggestion would be just place a single call & observe. See if it > happens of a single call. No, it is not TLS enabled, so probably not that. Also it is random, redial and it is fine... Happens about once every 10 or 20 calls. Thanks Richard >

1 0

Re: [Users] presence agent?
by Henri Keski-Sikkil� 18 Oct '06

18 Oct '06

On 17/Oct/2006 18:08 Cesc wrote .. > nice ... i will take a look. > BTW, any experience with phones? > > Cesc Hello, I have used x-lite, wengo-ng and presence server and it works well even though there are some presence bugs in wengo-ng... Best Regards, Henri Keski-Sikkila

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[Serusers] serctl ping uri error 408
by Ali Zaidi 18 Oct '06

18 Oct '06

Hi, Just wondering if someone is facing similar problem. My setup is SER/MediaProxy/MYSQL on public IP, I have couple of dozen SPA-1001 all over the internet. The problem I'm facing is that many of these ATA SPA-1001 some times do not ring, so try to ping (serctl ping sip:550505050@myserver.com) and i get error 408. When i restart these ATA everthing starts working but couple of hours later, it's the same problem. Not to mention that (serctl showdb) shows that sip client is registered, however I still get 408. I tried the following: (but no success) 1. SPA-1001 change Register Expires: from 3600 to 600 2. re-INVITE TO 30 Can someone help, or maybe there are some manufacturing bug on some linksys SPA-1001, by the way all my ATA use uniform configs and firmware. If someone is willing to look into the ngrap, i can show it to you. Thanks, Ali...

1 0

Re: [Users] presence agent?
by henri.keski-sikkila＠sesca.com 18 Oct '06

18 Oct '06

On 17/Oct/2006 18:08 Cesc wrote .. > nice ... i will take a look. > BTW, any experience with phones? > > Cesc Hello, I have used x-lite, wengo-ng and presence server and it works well even though there are some presence bugs in wengo-ng... Best Regards, Henri Keski-Sikkila

1 0

[Serusers] rtp-asymmetric-clients problem.
by Ricardo Martinez 18 Oct '06

18 Oct '06

Hello User. I using a rtp-asymmetric client in my platform, i want to add this client as a asymmetric client so i using this configuration. [ser.cfg] modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients") [rtp-asymmetric-clients file in /usr/local/etc/ser/] # # Lines starting with a '#' character or empty lines are ignored. # Put a User-Agent name or regular expression per line. # Check is case insensitive. # # This file should only list SIP clients that are asymmetric regarding # the RTP media streams. For clients that are asymmetric regarding the SIP # signaling, use the equivalent SIP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # SIPquest-SIPH323-Gateway/R2-23 When the 200 - OK from the asymmetric client arrive : SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.148.246;branch=z9hG4bK9eda.3d49eaf1.0. Via: SIP/2.0/UDP 192.168.1.100;received=100.100.148.248;rport=46107;branch=z9hG4bKc0a801640000004b453532800000783900000087. Record-Route: <sip:100.100.154.36;ftag=10302336931863;lr=on>. Record-Route: <sip:7072408196@100.100.148.246:5060;nat=yes;ftag=10302336931863;lr=on>. From: "unknown"<sip:5502408196@sip.desa.mydomain.net>;tag=10302336931863. To: <sip:7072408196@sip.desa.mydomain.net>;tag=3235014976. Call-ID: E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100. Cseq: 2 INVITE. Date: Tue, 17 Oct 2006 19:44:02 GMT. Server: SIPquest-SIPH323-Gateway/R2-23. Content-Type: application/sdp. Content-Length: 162. Contact: sip:7072408196@100.100.154.119. . v=0. o=100.100.154.119 1 627067186 IN IP4 100.100.154.119. s=SIP Library call. c=IN IP4 100.100.148.230. t=3370103042 0. m=audio 17322 RTP/AVP 0. a=rtpmap:0 PCMU/8000. Oct 17 16:45:18 sip mediaproxy[1783]: command lookup E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100 100.100.148.230:17322:audio 100.100.154.36 sip.desa.mydomain.net local sip.desa.mydomain.net unknown SIPquest-SIPH323-Gateway/R2-23 info=from:5502408196@sip.desa.mydomain.net,to:7072408196@sip.desa.mydomain.net,fromtag:10302336931863,totag:3235014976 Oct 17 16:45:18 sip mediaproxy[1783]: command execution time: 0.55 ms Oct 17 16:45:18 sip mediaproxy[1783]: session E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100: caller signed in from 100.100.148.248:49172 (RTP) (will return to 100.100.148.248:49172) Oct 17 16:45:18 sip mediaproxy[1783]: session E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100: called signed in from 100.100.148.230:17090 (RTP) (will return to 100.100.148.230:17090) There is no match with the rtp-asymmetric-clients file ? What i'm doing wrong? I hope that someone can help me here. Thanks Ricardo.-

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[Serusers] Re: [Serdev] pa module: presence "substatus"
by Vaclav Kubart 18 Oct '06

18 Oct '06

Hi, notes are quite often used for this. Can you send your config here? What SER version do you use? And if you try register your users via sending REGISTER messages it is still the same? Vaclav P.S. Send these messages to serusers too, might be that somebody else will have the same problem... On Tue, Oct 17, 2006 at 05:18:55PM +0200, Diego Do?ate wrote: > > Hi, > > Thanks, then I will use a <note> tag for the presence > substatus(<note>Away<\note>, for instance). > > I have a problem with the Subscription. User A subscribes to the presence > of user B and receives a NOTIFY but with "close/offline", and both users are > previously registered successfully in SER. > > Both users are added with serctl, but it seems the SER does not "save" > user A registration when user B subscribes... Where may be the problem? May > it be related to autherization? By now, it is disabled... > > Thanks in advance > > > > -----Mensaje original----- > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Enviado el: lunes, 16 de octubre de 2006 18:57 > Para: Diego Do?ate > CC: serdev(a)lists.iptel.org > Asunto: Re: [Serdev] pa module: presence "substatus" > > Hi, > support for PIDF extensions (elements from non-pidf namespace) is nearly > finished in current CVS version, but it still needs some work. It will be > done as soon as possible. > > Notes should be possible in last presence snapshot without any problems > - they can be within tuple or presence elements according specification. > > Vaclav > > On Mon, Oct 16, 2006 at 05:52:47PM +0200, Diego Do?ate wrote: > > > > Hi, > > > > In the SER "pa" module, I would like to use a tag like "note" or > > "substatus" in the pidf of the NOTIFY when a PUBLISH is received (with > > a presence change, to "Away", for instance), but SER only modified the > > <status> tag ("open"/"closed"). > > > > How can I implement a more detailed presence status management in > > the SER? > > > > Thanks in advance > > > > > > _______________________________________________ > > Serdev mailing list > > Serdev(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serdev >

1 0

[Serusers] serctl ping 408
by Ali Zaidi 18 Oct '06

18 Oct '06

Hi, Just wondering if someone is facing similar problem. My setup is SER/MediaProxy/MYSQL on public IP, I have couple of dozen SPA-1001 all over the internet. The problem I'm facing is that many of these ATA SPA-1001 some times do not ring, so try to ping (serctl ping sip:550505050@myserver.com) and i get error 408. When i restart these ATA everthing starts working but couple of hours later, it's the same problem. Not to mention that (serctl showdb) shows that sip client is registered, however I still get 408. I tried the following: (but no success) 1. SPA-1001 change Register Expires: from 3600 to 600 2. re-INVITE TO 30 Can someone help, or maybe there are some manufacturing bug on some linksys SPA-1001, by the way all my ATA use uniform configs and firmware. If someone is willing to look into the ngrap, i can show it to you. Thanks, Ali...

1 0

[Users] presence agent?
by Cesc 17 Oct '06

17 Oct '06

Hi, I saw some chatter a few days ago about openser's presence agent being about to be put on the CVS ... any idea on the ETA? Two students are coming to realize a project for a SIP-based content routing module, and the PA could be a good base. They should be using either ser's pa or openser's pa for processing the sub/not/pub messages, probably extending a bit the functionality ... And if the PA is already available (say ... 2 weeks), i can force it on them to use. Regards, Cesc

6 6

[Users] Syslog tutorial
by Marnus van Niekerk 17 Oct '06

17 Oct '06

2 1

[Users] calls being disconnected
by Richard Bennett 17 Oct '06

17 Oct '06

Hi, I am having increasing problems with calls being cut-off after a few seconds. I'm running openser with this version: version: openser 1.1.0-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. @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ main.c compiled on 04:14:03 Aug 11 2006 with gcc 4.1.1 with mediaproxy 1.7.2 running on the same dual cpu server. At the time of the disconnect there were 100 calls going over mediaproxy (cpu at 46%), but it also happens with lower load. The pstn gateway is a Lucent Max TNT. My config is based on the onsip gettingstarted examples, except that all calls are sent over mediaproxy, by setting use_media_proxy even if no nat problems are detected: if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); ## use_media_proxy(); }; use_media_proxy(); #use mediaproxy always and: #if (isflagset(6) || isflagset(7)) { # if (!isflagset(8)) { # setflag(8); # use_media_proxy(); # }; #}; if (!isflagset(8)) { setflag(8); use_media_proxy(); }; I can attach the full config if required. I had one problem when upgrading to openserv1.1,0, that this didn't work anymore: avp_write("i:45", "inv_timeout"); in the pstn route. So this is missing, and the module section has been changed: modparam("tm", "fr_inv_timer", 27) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") I tried finding the correct syntax from the latest documentation, but nothing worked. Could this be causing call disconnects? I attached a ngrep trace of a call that disconnected after a few seconds. It looks like my ua sends two ACKs to the gateway, doesn't receive a reply, and then send a bye, but I'm not sure why all that happens. Any help or pointers is appreciated. Richard

1 0

[Users] A question about acc flags..
by nick 17 Oct '06

17 Oct '06

I'm doing dfferent levels of accounting, radius for billing concerns, mysql for some further in depth analysis. What I'm wondering about is the numbers used in the modparam() lines, I know that the params for radius_flag, radius_missed_flag, etc need to be different, but can the numbers for radius_flag, db_flag and log_flag overlap? for example, all _flag acc would be one, all _missed_flag 2, etc etc?? I haven't seen anything that addressed the issue, either as a possibility or as something to avoid, do I need to worry about it? Nick

2 1

[Users] openserctl and permanent ul contacts
by Andreas Granig 17 Oct '06

17 Oct '06

Hi, I've some registrars and proxies using db_mode 3 on OpenSER 1.1.0. When I try to add a permanent contact into the location table using openserctl, the "permanent"-flag (128) is set, but the contact is removed upon hit of next timer. I quickly checked the source code, but couldn't find any timer handling for mode DB_ONLY, so maybe one of the core developers could take a look. Thanks, Andy

2 1

[Users] REGISTER problems
by Jon Schøpzinsky 17 Oct '06

17 Oct '06

Hello Im having a somewhat weird problem, at least for somebody like me, who is accustomed to the lovely NAT handling functionality of Asterisk. When my phone registers on our internal network, everything works like a charm, and the user is authenticated. But when I move the phone out on the public internet, it doesn't. I can see that the only thing changed in the sip messages, is our Cisco PIX firewall, that changes a few of the IP addresses. How can this have an impact on the calculation of the HA1 string, since this is where it fails. The realm hasn't changed, so I cant understand why this doesn't work. Could somebody who is a little more accustomed to OpenSER tell me why this happends? Med venlig hilsen Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefoni Tlf.: +45 70 20 40 59 Direkte: +45 88 20 03 36 Mobil: +45 31 20 67 09 http://www.detele.dk Dansk Erhvervs-Telefon er den mest fleksible og service-orienterede teleudbyder til dansk erhvervsliv. Derfor leverer vi til stadighed nye innovative og værdiskabende telefoniløsninger til danske virksomheder. Produkterne bygger på vores grundlæggende filosofi om, at overblik og service skal gå hånd i hånd med den rigtige pris. Hos Dansk Erhvervs-Telefoni, mener vi, teknologivalget skal styres af behovene hos den enkelte virksomhed. Vi tilbyder såvel løsninger baseret på Voice over IP (VoIP) som løsninger baseret på traditionel ISDN-teknologi. Vi mestrer teknologierne, men vores drivkraft er kundens behov. Det er vigtigst for os, at rådgive ud fra en behovsanalyse - ikke ud fra teknologifascination. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.4/476 - Release Date: 14-10-2006

3 2

[Users] Flag
by Gregoire 17 Oct '06

17 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! Does anyone have a list of the 32 flags that can be set in the configuration, and their signification? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFM4nHI8gmGeMTr0sRAuKMAJ9Y32PNwMyQwdYi+REYX30wnnl91ACdFhrg /DZnyUUakReZTObMgQuKU50= =TTZC -----END PGP SIGNATURE-----

2 1

[Users] Cisco authentication
by Marco F. Delaurenti 17 Oct '06

17 Oct '06

Hi all, I've a working openser and a cisco router (17xx). I'd like to authenticate the cisco router as a sip ua. If I disable www_authorize everything is working fine and the router register to the server. If I enable it and configure the router with sip-ua authentication username xxx password xxx realm xxx I see the router trying to register, but no way to make it working. I tried I think every combination of realm in www_authorize, www_challenge and so on. If I debug a successful authentication, I see a query on the db, "select ... from subscriber". If I debug the cisco auth, the server doesn't even try the query. Has somebody any hint please? TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain

2 1

[Serusers] How to Register SER server
by jk 17 Oct '06

17 Oct '06

Hello, May be its a very simple question, I have a SIP account from one VoIP service provider. username:password@domain-name-of-voip-service-provider. How can I register my ser server to receive calls. -JK

2 1

Re: [Users] polycom on hold, IN IP4 0.0.0.0 and force_rtp_proxy()
by Benko 17 Oct '06

17 Oct '06

On Tue, 17 Oct 2006 10:36:20 -0400 Nathan Hawkins <utsl(a)quic.net> wrote: > Comments inline. > My gut feeling > is you've found a firmware bug, or it's not interacting right with > the other phone. I'd try calling between two Polycom's and see if > that works as expected. > > Good luck! no, it doesn't make a difference, i just tried it with a polycom 601 and 501 involved. While i guess your assumption that this is incorrect behaviour of the polycom is true i'm not sure if this is the main reason for the one-way audio issue i initially posted. The sdp-part in the invite message on resuming the call on hold looks like it would have to correct information to go through the nat, the "c=a.b.c.d"-line is updated with the proxy ip and all other nat-parameters are appended too. Could this be a problem of rtpproxy or asterisk? The rtp-ports change a lot, seems some device sends the audio to a port that has not been recognized by the firewall. Hope someone has some additional input, i'll try to contact polycom but with their strict reseller policy i guess i'll not get an answer... best regards christian > > To clarify what i'm talking about i've attached the whole sip-debug > > of two calls, the first call is sent on hold when a second call > > comes and resumed after the second call has ended(1111111801 calls > > 2222222804 and it answers, then 0891234567 calls 2222222804 and > > 1111111801 is set on hold and resumed later). Openser runs on .98, > > which also serves the rtpproxy. .97 is a asterisk-gateway. > > > > Thanks for the input > > Christian > > > > > > > > > > > >> What firmware version are you running? > >> > >> Nathan > >> > >> Benko wrote: > >> > >>> Hmm, actually this parameter is set to 0(default). > >>> However, in practice it still uses the old standard(rfc2543). > >>> There's no media direction parameter in the sdp-message sent by > >>> the polycom for some reason, although the manual states to do so > >>> (firmware 2.0.1). Is it possible to force this somehow? > >>> > >>> regards > >>> christian > >>> > >>> On Sun, 15 Oct 2006 19:58:33 -0400 > >>> Nathan Hawkins <utsl(a)quic.net> wrote: > >>> > >>> > >>>> There's an option for the Polycom phones to switch the hold > >>>> behaviour. > >>>> > >>>> Set voIpProt.SIP.useRFC2543hold to 0, and it should use RFC3264 > >>>> rules for signalling hold instead of 0.0.0.0. > >>>> > >>>> Benko wrote: > >>>> > >>>>> Hello! > >>>>> > >>>>> I'm having a issue with NAT and rtpproxy. Usually my setup works > >>>>> fine with natted clients, the Connection Information is > >>>>> overwritten with the IP of the rtpproxy and audio passes through > >>>>> in both directions. However, today i came across a problem where > >>>>> the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the > >>>>> private ip after resuming a call that was on hold(actually, the > >>>>> other party is invited again) - and the force_rtp_proxy > >>>>> ()-command on openser left the ip untouched instead of > >>>>> overwriting it with the rtpproxy-ip. As a result the person that > >>>>> was on hold had audio but the polycom user (with the "wrong" ip) > >>>>> hadn't. > >>>>> > >>>>> The false ip left aside, is it expected behaviour of > >>>>> force_rtp_proxy to not touch 0.0.0.0? > >>>>> > >>>>> Just out of curiosity - does someone know the "on hold"-problem > >>>>> with polycoms? > >>>>> > >>>>> thx > >>>>> christian > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users(a)openser.org > >>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>> > >> ------------------------------------------------------------------------ > >> > >> interface: bond0 (2XX.XXX.XXX.96/255.255.255.240) > >> filter: ip and ( port 5060 and host 81.XXX.XXX.196 ) > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 INVITE. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> P-Key-Flags: keys="3". > >> User-Agent: snom190/3.60w. > >> Accept: application/sdp. > >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > >> SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. > >> Supported: timer, 100rel, replaces. > >> Session-Expires: 3600. > >> Content-Type: application/sdp. > >> Content-Length: 268. > >> . > >> v=0. > >> o=root 1814496751 1814496751 IN IP4 10.1.99.169. > >> s=call. > >> c=IN IP4 10.1.99.169. > >> t=0 0. > >> m=audio 55428 RTP/AVP 8 0 18 101. > >> a=rtpmap:8 pcma/8000. > >> a=rtpmap:0 pcmu/8000. > >> a=rtpmap:18 g729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-15. > >> a=ptime:20. > >> a=sendrecv. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> SIP/2.0 407 Proxy Authentication Required. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport=2051;received=81.XXX.XXX.196. > >> From: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=2f7d8fa65b1e4a32386804888f034ca8.5f9f. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 INVITE. > >> Proxy-Authenticate: Digest realm="sip.de.mydomain.org", > >> nonce="4534dc22e41f8b3a51321f458eed7830772d2ea7". Server: OpenSer > >> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392 > >> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907 > >> req_src_ip=81.XXX.XXX.196 req_src_port=2051 > >> in_uri=sip:0892222222804@sip.de.mydomain.org;user=phone > >> out_uri=sip:0892222222804@sip.de.mydomain.org;user=phone > >> via_cnt==1". . > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=2f7d8fa65b1e4a32386804888f034ca8.5f9f. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 ACK. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-x9wr16oqr17g;rport. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> P-Key-Flags: keys="3". > >> User-Agent: snom190/3.60w. > >> Accept: application/sdp. > >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > >> SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. > >> Supported: timer, 100rel, replaces. > >> Session-Expires: 3600. > >> Proxy-Authorization: Digest > >> username="11111111801",realm="sip.de.mydomain.org",nonce="4534dc22e41f8b3a51321f458eed7830772d2ea7",uri="sip:0892222222804@sip.de.mydomain.org;user=phone",response="139c40e2fecbadf4412628412607a7d7",algorithm=md5. > >> Content-Type: application/sdp. Content-Length: 268. > >> . > >> v=0. > >> o=root 1814496751 1814496751 IN IP4 10.1.99.169. > >> s=call. > >> c=IN IP4 10.1.99.169. > >> t=0 0. > >> m=audio 55428 RTP/AVP 8 0 18 101. > >> a=rtpmap:8 pcma/8000. > >> a=rtpmap:0 pcmu/8000. > >> a=rtpmap:18 g729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-15. > >> a=ptime:20. > >> a=sendrecv. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> SIP/2.0 100 trying -- your call is important to us. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-x9wr16oqr17g;rport=2051;received=81.XXX.XXX.196. > >> From: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> To: <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE. > >> Server: OpenSer (1.1.0-tls (i386/linux)). > >> Content-Length: 0. > >> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20909 > >> req_src_ip=81.XXX.XXX.196 req_src_port=2051 > >> in_uri=sip:0892222222804@sip.de.mydomain.org;user=phone > >> out_uri=sip:0892222222804@mgw.de.mydomain.org;user=phone > >> via_cnt==1". . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;received=81.XXX.XXX.196;branch=z9hG4bK-x9wr16oqr17g;rport=2051. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=lxlk4gyz7h>. From: > >> "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE. > >> User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0892222222804@2XX.XXX.XXX.97>. > >> Content-Type: application/sdp. > >> Content-Length: 303. > >> . > >> v=0. > >> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36578 RTP/AVP 0 18 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=nortpproxy:yes. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> INVITE sip:2222222804@10.1.99.161:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>. Contact: > >> <sip:0891111111@2XX.XXX.XXX.97:5060>. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. CSeq: 102 > >> INVITE. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Date: Tue, 17 Oct 2006 13:30:30 GMT. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Content-Type: application/sdp. > >> Content-Length: 396. > >> . > >> v=0. > >> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36580 RTP/AVP 0 18 4 3 97 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:4 G723/8000. > >> a=rtpmap:3 GSM/8000. > >> a=rtpmap:97 iLBC/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=direction:active. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0892222222804@2XX.XXX.XXX.97 SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-kqgbv28tcm8k;rport. Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=lxlk4gyz7h>. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 ACK. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 100 Trying. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. CSeq: > >> 102 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. CSeq: > >> 102 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. Allow: INVITE, ACK, > >> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, > >> UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Type: > >> application/sdp. Content-Length: 185. . > >> v=0. > >> o=- 1161091665 1161091665 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> m=audio 2266 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> ACK sip:2222222804@81.XXX.XXX.196:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.2. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK77d83684;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. CSeq: 102 > >> ACK. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Content-Length: 0. > >> . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> INVITE sip:2222222804@10.1.99.161:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>. Contact: > >> <sip:0891234567@2XX.XXX.XXX.97:5060>. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 102 > >> INVITE. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Date: Tue, 17 Oct 2006 13:30:38 GMT. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Content-Type: application/sdp. > >> Content-Length: 396. > >> . > >> v=0. > >> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36584 RTP/AVP 0 18 4 3 97 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:4 G723/8000. > >> a=rtpmap:3 GSM/8000. > >> a=rtpmap:97 iLBC/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=direction:active. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 100 Trying. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> CSeq: 102 INVITE. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 180 Ringing. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> CSeq: 102 INVITE. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Allow-Events: > >> talk,hold,conference. Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 1 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Max-Forwards: 70. > >> Content-Type: application/sdp. > >> Content-Length: 181. > >> . > >> v=0. > >> o=- 1161091665 1161091666 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 0.0.0.0. > >> t=0 0. > >> m=audio 2266 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 407 Proxy Authentication Required. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 1 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Proxy-Authenticate: Digest realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b". Server: OpenSer > >> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392 > >> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 1 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Max-Forwards: 70. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKf40b74c7C2612792. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 2 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="a690d1208542f7132ef16747f0abb98a", algorithm=MD5. > >> Max-Forwards: 70. Content-Type: application/sdp. Content-Length: > >> 181. . v=0. o=- 1161091665 1161091666 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 0.0.0.0. > >> t=0 0. > >> m=audio 2266 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 100 trying -- your call is important to us. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bKf40b74c7C2612792;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 2 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Server: > >> OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0. > >> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;rport=5060;received=81.XXX.XXX.196;branch=z9hG4bKf40b74c7C2612792. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> CSeq: 2 INVITE. User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. > >> Content-Type: application/sdp. > >> Content-Length: 232. > >> . > >> v=0. > >> o=root 6068 6069 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36580 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK4ec65e93B39DC51E. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 2 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Proxy-Authorization: > >> Digest username="2222222804", realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="a690d1208542f7132ef16747f0abb98a", algorithm=MD5. > >> Max-Forwards: 70. Content-Length: 0. . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> CSeq: 102 INVITE. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. Allow: INVITE, ACK, > >> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, > >> UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Type: > >> application/sdp. Content-Length: 185. . > >> v=0. > >> o=- 1161091674 1161091674 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> m=audio 2268 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> ACK sip:2222222804@81.XXX.XXX.196:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.2. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK4365390f;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> Contact: <sip:0891234567@2XX.XXX.XXX.97>. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 102 > >> ACK. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> BYE sip:0891234567@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKc2fe28198F44BA84. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> To: "0891234567" > >> <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. CSeq: 1 BYE. > >> Call-ID: 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. > >> Max-Forwards: 70. > >> Content-Length: 0. > >> . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;received=81.XXX.XXX.196;branch=z9hG4bKc2fe28198F44BA84. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=4245106-795C21FB>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> To: "0891234567" > >> <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 1 BYE. > >> User-Agent: MyMediaGW. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, > >> REFER, SUBSCRIBE, NOTIFY. Contact: <sip:0891234567@2XX.XXX.XXX.97>. > >> Content-Length: 0. > >> X-Asterisk-HangupCause: Normal Clearing. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 3 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="4fafb927667083777a520a6f717fc8d3", algorithm=MD5. > >> Max-Forwards: 70. Content-Type: application/sdp. Content-Length: > >> 247. . v=0. o=- 1161091665 1161091667 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> a=sendrecv. > >> m=audio 2266 RTP/AVP 0 8 18 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:18 G729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 407 Proxy Authentication Required. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 3 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Proxy-Authenticate: Digest realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932". Server: OpenSer > >> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392 > >> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20903 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 3 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Max-Forwards: 70. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK3fa5014e6D19C349. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 4 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="e6f5f00560921aceca9a0329333b720f", algorithm=MD5. > >> Max-Forwards: 70. Content-Type: application/sdp. Content-Length: > >> 247. . v=0. o=- 1161091665 1161091667 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> a=sendrecv. > >> m=audio 2266 RTP/AVP 0 8 18 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:18 G729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 100 trying -- your call is important to us. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bK3fa5014e6D19C349;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 4 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Server: > >> OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0. > >> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20903 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;rport=5060;received=81.XXX.XXX.196;branch=z9hG4bK3fa5014e6D19C349. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> CSeq: 4 INVITE. User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. > >> Content-Type: application/sdp. > >> Content-Length: 303. > >> . > >> v=0. > >> o=root 6068 6070 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36580 RTP/AVP 0 18 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKca5c1daDE079395. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 4 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Proxy-Authorization: > >> Digest username="2222222804", realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="e6f5f00560921aceca9a0329333b720f", algorithm=MD5. > >> Max-Forwards: 70. Content-Length: 0. . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> BYE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKcda270c0BA153C5B. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 5 BYE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="d1e585adb491da5b0643fcdb7a5d8cef", algorithm=MD5. > >> Max-Forwards: 70. Content-Length: 0. . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;received=81.XXX.XXX.196;branch=z9hG4bKcda270c0BA153C5B. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> CSeq: 5 BYE. User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. > >> Content-Length: 0. > >> X-Asterisk-HangupCause: Normal Clearing. > >> . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> BYE sip:11111111801@81.XXX.XXX.196:2051 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as0fd69c4e>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bKf278.897153c6.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK13f9319f;rport=5060. From: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> To: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> Contact: <sip:0892222222804@2XX.XXX.XXX.97>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 102 BYE. > >> User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bKf278.897153c6.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK13f9319f;rport=5060. From: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> To: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 102 BYE. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> User-Agent: snom190/3.60w. > >> Content-Length: 0. > >> . > >> > >> exit > >> 78 received, 0 dropped > >> >

1 0

[Serusers] route logic
by vijay tiwari 17 Oct '06

17 Oct '06

hi all i have facing a problem that i am not able to send the call on other sip proxy where in route logic give the ip address of particullar sip proxy to send the call on it. thanks vijay --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

1 0

Re: [Users] polycom on hold, IN IP4 0.0.0.0 and force_rtp_proxy()
by Benko 17 Oct '06

17 Oct '06

On Tue, 17 Oct 2006 08:08:32 -0400 Nathan Hawkins <utsl(a)quic.net> wrote: > I'm testing Polycom 601's with 2.0.1 firmware and Mediaproxy. I > haven't noticed any problems with hold at all, so I haven't looked > into whether it's using 0.0.0.0 or not. I supposed it's possible that > that parameter doesn't actually work, and Mediaproxy just works with > it. > > I think though, that what you're describing below, where it resumes > from hold with 0.0.0.0 as the address, is a firmware bug. The Polycom > should definitely not be doing that. That's going to break all kinds > of things. hmm, guess i missunderstood - should the media direction parameter also be marked when "on hold" is "started" or only at the end when we resume the call that was on hold? Currently, Polycom sets IP 0.0.0.0 at the beginning of the hold, no media path direction is set. When the call is resumed later, the polycom sets the correct "sendrecv"-parameter (Which i didn't recognize earlier, sorry). However, if i understood rfc3264 the polycom should mark the media stream as sendonly on the beginning of the hold... To clarify what i'm talking about i've attached the whole sip-debug of two calls, the first call is sent on hold when a second call comes and resumed after the second call has ended(1111111801 calls 2222222804 and it answers, then 0891234567 calls 2222222804 and 1111111801 is set on hold and resumed later). Openser runs on .98, which also serves the rtpproxy. .97 is a asterisk-gateway. Thanks for the input Christian > > What firmware version are you running? > > Nathan > > Benko wrote: > > Hmm, actually this parameter is set to 0(default). > > However, in practice it still uses the old standard(rfc2543). > > There's no media direction parameter in the sdp-message sent by the > > polycom for some reason, although the manual states to do so > > (firmware 2.0.1). Is it possible to force this somehow? > > > > regards > > christian > > > > On Sun, 15 Oct 2006 19:58:33 -0400 > > Nathan Hawkins <utsl(a)quic.net> wrote: > > > >> There's an option for the Polycom phones to switch the hold > >> behaviour. > >> > >> Set voIpProt.SIP.useRFC2543hold to 0, and it should use RFC3264 > >> rules for signalling hold instead of 0.0.0.0. > >> > >> Benko wrote: > >>> Hello! > >>> > >>> I'm having a issue with NAT and rtpproxy. Usually my setup works > >>> fine with natted clients, the Connection Information is > >>> overwritten with the IP of the rtpproxy and audio passes through > >>> in both directions. However, today i came across a problem where > >>> the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the > >>> private ip after resuming a call that was on hold(actually, the > >>> other party is invited again) - and the force_rtp_proxy > >>> ()-command on openser left the ip untouched instead of > >>> overwriting it with the rtpproxy-ip. As a result the person that > >>> was on hold had audio but the polycom user (with the "wrong" ip) > >>> hadn't. > >>> > >>> The false ip left aside, is it expected behaviour of > >>> force_rtp_proxy to not touch 0.0.0.0? > >>> > >>> Just out of curiosity - does someone know the "on hold"-problem > >>> with polycoms? > >>> > >>> thx > >>> christian > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Users] Dual phone (wifi SIP/GSM) compatible list ?
by Christophe Irles 17 Oct '06

17 Oct '06

Hi, Do you know dual phones (wifi SIP and GSM) compatible with openser ? For example is Nokia E61 compatible ? Any others ? Thanks in advance, Christophe

4 3

[Users] polycom on hold, IN IP4 0.0.0.0 and force_rtp_proxy()
by Benko 17 Oct '06

17 Oct '06

Hello! I'm having a issue with NAT and rtpproxy. Usually my setup works fine with natted clients, the Connection Information is overwritten with the IP of the rtpproxy and audio passes through in both directions. However, today i came across a problem where the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the private ip after resuming a call that was on hold(actually, the other party is invited again) - and the force_rtp_proxy ()-command on openser left the ip untouched instead of overwriting it with the rtpproxy-ip. As a result the person that was on hold had audio but the polycom user(with the "wrong" ip) hadn't. The false ip left aside, is it expected behaviour of force_rtp_proxy to not touch 0.0.0.0? Just out of curiosity - does someone know the "on hold"-problem with polycoms? thx christian

3 4

[Serusers] How-to write a custom module
by Steve Blair 17 Oct '06

17 Oct '06

Samuel: A while back you responded to one of my posts about writing modules for SER. I am currently writing a module and I have an issue I do not understand. I'm hoping you can help. I created a directory under the ser-0.9.7-pre3/modules source directory on my server and placed my code there along with a Makefile. If I make the code from this directory it compiles with two warnings including one about running make from the main directory. If I cd to the main directory and run make clean my new subdirectory is visited as expected. If I then run make my modules does not get compiled. I believe the new directory is not recognized but I cannot figure out why. Would you hapen to be able to explain this behavior? Thank you, Steve

3 2

RE: [Serusers] :Check for URI (from incoming messages) w.r.t database
by Kamal.Mann＠t-systems.com 17 Oct '06

17 Oct '06

Got my answer Regards Kamal Mann -----Original Message----- From: Mann, Kamal Sent: Tuesday, October 17, 2006 11:37 AM To: serusers(a)lists.iptel.org Subject: [Serdev] :Check for URI (from incoming messages) w.r.t database Hi All I need to check for each URI with respect to his/her credentials stored in SER database. I am using avpops functions avp_check & avp_db_load. I am able to compile ser.cfg but after writing this piece of code ser stops responding to all kinda requests not even for register messages. Like --> if(avp_check("s:kamal","eq/$from/username")) //avp_db_load("$from/username","s:kamal") // also tried with scheme1: example { ****some more code**** break; } Is this way is correct or I need to use some another function?? I am a newbie to ser; I need your valuable suggestions for the same. Regards Kamal Mann

1 0

[Serusers] Authenticate when routing calls to SIP-gateway. How?
by Evert 17 Oct '06

17 Oct '06

Hi all! I am experimenting a bit with SER, and am trying to get it to do some of the same things I now have Asterisk for. But I am having trouble with the following: I want to route all non-local calls to my SIP service provider. But for this to work I need to authenticate myself first towards my SIP service provider. How do I do this with SER? Regards, Evert

2 2

Re: [Serusers] SIP to AS communication
by Jiri Kuthan 17 Oct '06

17 Oct '06

Just a warning small in size but I guess big in the impact. I think people kind of underappreciate the effort which has to be taken. If you consider all the 3gpp complexity (sigcomp as a prime example, one of many being real non-trivial), size of mobile networks for which the system has to be running, number of interfaces to speak over and achieve interoperability (management, diameter, applications, cdr processing...), routing features (the 'scim' feature with which this discussion began)... I occur to know Tekelec has gone through this exercise and generating a reasonable result has cost quite an investment. At 07:19 06/10/2006, Andrey Kuprianov wrote: >Hi, I suppose you need to develop modules for all the the x-CSCF + AS (where x-CSCF is P-CSCF, S-CSCF or I-CSCF). You see, the functionalities/features that x-CSCF and application servers require (for instance, P-headers or querying for subscriber information from HSS) are not available in SER. That's why you need to make your own modules *all the way*. As for HSS, i dont think you need to make some special super DIAMETER-enabled database. In my (an in many other people's btw) point of view a simple database, like MySQL is more than capable of storing subscriber info or initial filtering criterias (iFC --> im talking about SCIM/SIMF). If you are really set to develop IMS on SER, then maybe you can start with SER 0.8.14. You can still find decent development documentation for it. As for later versions, I dont think any good docs are yet available (someone, please, correct me if im wrong) The documentation is being worked on, but for the moment you are unfortunately correct. >Just for your info, INT (France) joint with France Telecom are developing IMS on SER as of now (dont know if it's going to be commerial or open-source. >Contact them if you are interested, maybe you can even help them develop it further :). Plus, Focus is also set to develop a commercial version of IMS for SER and, although, Focus said that their IMS will be available in the last quarter of this year, but nothing's been released yet. :( That's fantastic news, I'm looking forward to GPL contributions from these very respectable institutions. This is really a big piece of work and I am enormously positive about every single new piece of the puzzle. -jiri > Good luck, Andrey. On 10/6/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > Hi All > We are using SER as CSCF which communicates with HSS and Application Server. > ï§ Whenever a UA sends a request to SER then it should go to some specific table and look for services assigned to that particular UA. > ï§ If UA is authorize to use that service than SER send this to AS. > ï§ AS then send this request/response to Serving SER (S-CSCF) which further serves that UA. > I need your suggestions for the same that what kind of module we need to develop for this scenario or any valuable suggestions. > > Thanks in anticipation > Kamal Mann > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > </x-flowed> >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] if power failure happens in UA while call is in progress, Ser accounts wrong
by Ricardo Carvalho 17 Oct '06

17 Oct '06

Meanwhile I was making my scripts for management of the accounting, I realized and after some call has been answered if the called party simply has a power failure or network failure, no BYE message ever gets returned to the other party, what results in the acc table in one ACK message and after that no more messages belonging to that call-id. This can mean erroneously that the call is still in progress eternally, resulting in an enormous billing to the caller!! This is a very serious issue! How can I protect my system from this kind of problems? Thanks, Ricardo.

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[Serusers] udp_rcv_loop error
by jorain 17 Oct '06

17 Oct '06

Hi all, I get this error when running ser, -------------------------------------------------------------------------------- ERROR: udp_rcv_loop:recvfrom:[11] Resource temporarily unavailable -------------------------------------------------------------------------------- I search for this error and get this answer from the mailing list but i don't know how to solve it. -------------------------------------------------------------------------------- This error (EAGAIN) appears also when and udp packet with bad checksum is received. In this case the linux kernel will return EAGAIN. I think this is the case for you. Probably your "new" phone is placed before a very bad link which corrupts the packet (or you have a bad firewall) -------------------------------------------------------------------------------- Could you please tell me how to solve it? Thanks. Best Regards, jorain

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[Serusers] :Check for URI (from incoming messages) w.r.t database
by Kamal.Mann＠t-systems.com 17 Oct '06

17 Oct '06

Hi All I need to check for each URI with respect to his/her credentials stored in SER database. I am using avpops functions avp_check & avp_db_load. I am able to compile ser.cfg but after writing this piece of code ser stops responding to all kinda requests not even for register messages. Like --> if(avp_check("s:kamal","eq/$from/username")) //avp_db_load("$from/username","s:kamal") // also tried with scheme1: example { ****some more code**** break; } Is this way is correct or I need to use some another function?? I am a newbie to ser; I need your valuable suggestions for the same. Regards Kamal Mann

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[Serusers] [Serdev] :Check for URI (from incoming messages) w.r.t database
by Kamal.Mann＠t-systems.com 17 Oct '06

17 Oct '06

Hi All I need to check for each URI with respect to his/her credentials stored in SER database. I am using avpops functions avp_check & avp_db_load. I am able to compile ser.cfg but after writing this piece of code ser stops responding to all kinda requests not even for register messages. Like --> if(avp_check("s:kamal","eq/$from/username")) //avp_db_load("$from/username","s:kamal") // also tried with scheme1: example { ****some more code**** break; } Is this way is correct or I need to use some another function?? I am a newbie to ser; I need your valuable suggestions for the same. Regards Kamal Mann

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[Serusers] i have install ser but facing some problem
by vijay tiwari 16 Oct '06

16 Oct '06

hi , i have softphone(eyebeam) and i wnat send the call thourgh ser what r the changes i have do in my ser and i want just send the call form my ser to other sip proxy what the change i have to do. thanks vijay --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

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[Users] OpenSER config help to talk to PSTN..
by Nick De Cristofaro 16 Oct '06

16 Oct '06

Hello, I'm an OpenSER newbie and I realise this is probably a very fundemantal thing but I followed the installation tutorial and built some test users on a local MySQL database. Users can register correctly to the SER but they cannot call any other SIP users on the internet or anything else. I am not familiar with the architecture but what is required here? I have access to a Audiocodes Mediant 2000 and would like to get users to be able to call real PSTN numbers. What is required for me to-do to configure this? Many thanks! Nick

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[Serusers] SER/SEMS Server Appliances
by Goran Donev 16 Oct '06

16 Oct '06

I was wondering if anyone out there new of any company that makes SER/SEM Server appliances. The combination of SER/SEMS, make it a more viable solution then Asterisk. Awaiting for responses. Thanks ______________ Protected by MailSword. "Spam Ends Here" FREE download at: http://www.mailsword.com

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[Serusers] Ser and Syslog
by Ricardo Carvalho 16 Oct '06

16 Oct '06

Hi all, I'm trying to enable Ser to write log and debug messages to syslog, but I'm not having success... I'm using Ser-0.9.6 and in my ser.cfg script I have added the following entries: log_facility=LOG_LOCAL6 loadmodule "/usr/local/lib/ser/modules/xlog.so" And in among the routes code I've added lines like this: xlog("L_INFO", "request method: [$rm], from URI: [$fu], To URI: [$tu]"); I've added to the file /etc/syslog.conf the following line: local6.* /var/log/voip.log But nothing gets written to that file, not even to /var/log/messages file! Although, if I make log_stderror=yes in ser.cfg, all log and debug messages are shown just like was expected to the sterror! What am I doing wrong? What is lacking? Regards, Ricardo.

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[Users] Double Registration with Openser proxy and Openser registrar
by Lindsey Hans Joseph Lao 15 Oct '06

15 Oct '06

Hello, I currently have this setup client --> (private interface) OpenSER (public interface) --> registrar (public) In this scenario, I am configuring OpenSER to function as an application level gateway and at the same time a proxy server. I was able to successfully manipulate the contact headers thru textops module such a way that the IP registering in the registrar is the public one. NAThelper's fix_nated_contact and fix_nated_register won't work. No idea why... What I want to happen is that registration requests from the client would register with OpenSER (save in userlocation) and also register with the registrar, is there a way to do this? I'm using this script in the my proxy server such that if the destination uri is not intended for my proxy, it would substitute the contact headers and save it in "location" then relay the register request to its destination: lookup("aliases"); if(uri != myself) { if(is_method("REGISTER")) { append_hf("P-hint: Registration applied\r\n"); subst('/Contact:(.*)<sip:(.*)@(.*)>/Contact:<sip:\1@10.3.2.38:5060>/ig'); save("location"); t_relay(); exit; }; My problem is I am able to log in (i'm using xlite softphone) but upon checking the sip messages, it shows an 401 unauthorized message response from the server. Can anyone help me on this? Thanks much! Hans

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[Serusers] Compiling SER on Solaris 9
by Cheez Mirza 15 Oct '06

15 Oct '06

To compile SER on Solaris I ran (g)make all but get the following messages: Makefile.rules:84: action.d: No such file or directory Makefile.rules:84: crc.d: No such file or directory Makefile.rules:84: daemonize.d: No such file or directory Makefile.rules:84: data_lump.d: No such file or directory Makefile.rules:84: data_lump_rpl.d: No such file or directory Makefile.rules:84: dprint.d: No such file or directory Makefile.rules:84: dset.d: No such file or directory Makefile.rules:84: error.d: No such file or directory Makefile.rules:84: fifo_server.d: No such file or directory .... flex cfg.lex flex: fatal internal error, exec failed make: *** [lex.yy.c] Broken Pipe make: *** Deleting file `lex.yy.c' What am I doing wrong? Thanks, -CM __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] key exchange failed with MIKEY
by Gregoire 14 Oct '06

14 Oct '06

Hi everybody! I set up my Openser with TLS for signaling and it works well. I use Minisip as softphone and it registers successufully over TLS. But I have the following problem when I try to make a call using SRTP: Openser denies the INVITE cause its size is more than 2048 and then it makes an error as "Message too big". Even if I try to bypass this error in the configuration, the INVITE is still denied. In fact, I saw that Minisip uses MIKEY for the key exchange, and put it in the SDP body, and then it makes quite big but I don't understand why the size must be under 2048. Does someone have any idea to help me? Thanks Greg

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[Users] store in DB some searched content present in messages
by Ricardo Carvalho 14 Oct '06

14 Oct '06

Is there a way to search and store in a regular expression basis, some content of some kind of messages in an avp and then store that content in some table of MySQL database? Thanks, Ricardo.

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Re: [Users] acc_radius question
by Cseke Tamas 14 Oct '06

14 Oct '06

Hi, i use both setfalg(x), and acc_rad_request i need accounting for INVITE, ACK (for INVITE), CANCEL, BYE it was working ubkess error, but now i try failed_transaction flag. Could you help me in some better solution, I need acc_rad_request becasue of i need invite accounting immediately - for set the call setuptime ack - for set the call connecttime bye - for disconnecttime So i done it this way: (config in question) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "failed_transaction_flag", 4) if (is_method("ACK") || is_method("BYE")) { setflag(1); setflag(2); setflag(4); }; # accounting requests if (is_method("INVITE")) { if (!radius_www_authorize("$rd")) { www_challenge("$rd", "0"); exit; }; acc_rad_request("200 Invite received..."); consume_credentials(); } else if (is_method("ACK")) { if (t_check_status("[4-6][0-9][0-9]")) { acc_rad_request("Failed"); } else { acc_rad_request("200 Ack received..."); }; } else if (is_method("BYE")) { } else if (is_method("CANCEL")) { acc_rad_request("200 CANCEL received..."); }; Thanks, Tamas raviprakash sunkara wrote: > Can U send the Config > > U didn't set the flags properly in ur Config file > and > Don't use setflag("x"); > > Use acc_rad_request("200"); > > On 10/11/06, Cseke Tamas <cseke.tamas(a)eworldcom.hu> wrote: > >> >> Hi, >> >> When sends openser accounting packets for INVITE, and BYE message? >> it seems to be, when received 200 OK. >> >> 1)for INVITE i need radius packet immediately, not just when it was >> answered. >> 2)for BYE i need accounting packet, when any error occured too. >> >> modparam("acc", "radius_flag", 2) >> >> modparam("acc", "radius_missed_flag", 3) >> >> this attribute mean that, should sent radius packet, when any error >> occured too, isn't it? However openser don't send accounting packet in >> this case. >> >> i run into the following error: >> >> ########## >> T called -> openser >> BYE sip:21.16.8.92:5060 >> ## >> T openser->called >> SIP/2.0 477 Unfortunately error on sending to next hop occurred >> (477/TM) >> ########## >> T caller->openser >> BYE sip:66610038641368254@21.16.8.90:5060; >> ##### >> # openser -> caller >> SIP/2.0 481 Call Leg/Transaction Does Not Exist.. >> >> So no 200 OK for BYE received, therefore no accounting packet to radius >> sent. >> How can i solve these problems? >> >> Btw, what could cause this 477 error? >> >> Thanks, >> Tamas >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > > >

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Re: [Users] Please help me to fix openser.cfg (rewritehostport)
by Daniel-Constantin Mierla 14 Oct '06

14 Oct '06

Please keep the list in cc. use xlog("L_ERR", "$rm is next to pass for transfer to Asterisk"); maybe you got the log message for another request, the syntax looks ok. Cheers, Daniel On 10/13/06 14:38, Ronald Wiplinger wrote: > Daniel-Constantin Mierla wrote: >> I am not sure what do you want to achieve, but I assume it should be >> before lookup("aliases"); > > I want that if I dial a number, like 601, it should go to another > machine, where Asterisk is running and a phone with 601 is connected. > > I have moved the paragraph just before the lookup("aliases"); > save("location"); > exit; > }; > > xlog("L_ERR", "INVITE is next to pass > for transfer to Asterisk"); > if (method=="INVITE") { > xlog("L_ERR", "INVITE passed for > transfer to Asterisk"); > if (uri=~"^sip:601@") { > rewritehostport("59.124.210.35:5060"); > xlog("L_ERR", "transfer to Asterisk"); > route(1); > exit; > }; > }; > > lookup("aliases"); > > With these three xlog statements, I get only the first one (INVITE is > next to pass for transfer to Asterisk). The next if is obviously not > fulfilled! > Do I have a syntax error here? > > > bye > > Ronald Wiplinger > > >> >> Cheers, >> Daniel >> >> >> On 10/12/06 17:01, Ronald Wiplinger wrote: >>> I try to rewritehostport to my Asterisk box with all starting >>> numbers with 6. The xlog never shows anything. >>> >>> >>> >>> if (method=="INVITE") { >>> if (uri=~"^sip:6.+@") { >>> rewritehostport("xx.xx.xx.xx:5060"); >>> xlog("L_ERR", "transfer to Asterisk"); >>> route(1); >>> exit; >>> }; >>> }; >>> >>> I believe I did not put it at the right place. Below is my entire >>> openser.cfg. Can you please tell me, where above paragraph should >>> be? Other parts maybe also not in the right order. >>> >>> >>> >>> bye >>> >>> Ronald Wiplinger >>> >>> >>> >>> >>> >>> # >>> # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ >>> # >>> >>> debug=3 # debug level (cmd line: -dddddddddd) >>> fork=yes >>> log_stderror=no # (cmd line: -E) >>> log_facility=LOG_LOCAL7 >>> log_name="Taipei-openser" >>> >>> check_via=no # (cmd. line: -v) >>> dns=no # was no (cmd. line: -r) >>> rev_dns=no # (cmd. line: -R) >>> port=5060 >>> children=4 >>> fifo="/tmp/openser_fifo" >>> >>> >>> mpath="/usr/local/lib64/openser/modules/" >>> loadmodule "mysql.so" >>> loadmodule "sl.so" >>> loadmodule "tm.so" >>> loadmodule "rr.so" >>> loadmodule "maxfwd.so" >>> loadmodule "usrloc.so" >>> loadmodule "registrar.so" >>> loadmodule "textops.so" >>> loadmodule "nathelper.so" >>> loadmodule "xlog.so" >>> loadmodule "enum.so" >>> loadmodule "acc.so" >>> loadmodule "auth.so" >>> loadmodule "auth_db.so" >>> modparam("usrloc|auth_db","db_url","mysql://shshshsh:shhhhrw@localhost/wodeopenser") >>> modparam("usrloc", "db_mode", 2) >>> modparam("registrar", "nat_flag", 6) >>> modparam("registrar", "sip_natping_flag", 7) modparam("auth_db", >>> "calculate_ha1", yes) >>> modparam("auth_db", "password_column", "password") >>> modparam("rr", "enable_full_lr", 1) >>> modparam("nathelper", "rtpproxy_sock", "udp:xx.xx.xx.xx:22222") # >>> default port 22222 >>> #modparam("nathelper", "rtpproxy_sock", >>> "unix:/var/run/rtpproxy.sock") modparam("nathelper", >>> "natping_interval", 30) >>> modparam("nathelper", "ping_nated_only", 1) >>> modparam("nathelper", "sipping_from", "sip:xx.xx.xx.xx") >>> modparam("enum", "domain_suffix", "e164.org.") >>> modparam("enum", "tel_uri_params", ";npdi") >>> >>> >>> # ------------------------- request routing logic ------------------- >>> >>> # main routing logic >>> >>> route{ >>> xlog("L_ERR", "request method: [$rm], from URI: [$fu], To URI: >>> [$tu]"); >>> if (!mf_process_maxfwd_header("10")) { >>> sl_send_reply("483","Too Many Hops"); >>> exit; >>> }; >>> >>> if (msg:len >= 4096 ) { >>> sl_send_reply("513", "Message too big"); >>> exit; >>> }; >>> >>> # >>> ----------------------------------------------------------------- >>> # NAT detection >>> # >>> ----------------------------------------------------------------- >>> route(2); >>> >>> >>> # >>> ----------------------------------------------------------------- >>> # if you want to make ENUM work with numbers starting with "00", >>> # use the following to convert "00" it into a "+" >>> # >>> ----------------------------------------------------------------- >>> >>> >>> if (uri=~"^sip:00[1-9][0-9]*@") { strip(2); >>> prefix("+"); >>> }; >>> if (uri=~"sip:\+[0-9]+@") { >>> if(!enum_query("e164.arpa.") && !enum_query("e164.org.") && >>> !enum_query("e164.info.") && !enum_query("enum.org.")) >>> { >>> xlog("L_ERR", "Invalid enum? $uri >>> Route to PSTN"); >>> }; >>> xlog("L_ERR","we matched just number $tu"); >>> strip(1); # remove + >>> t_relay(); >>> exit; >>> }; >>> # >>> ----------------------------------------------------------------- >>> if (!method=="REGISTER") >>> record_route(); >>> >>> if (loose_route()) { >>> # mark routing logic in request >>> append_hf("P-hint: rr-enforced\r\n"); route(1); >>> }; >>> >>> # ----------------------------------------------------------------- >>> # Call Type Processing Section >>> # ----------------------------------------------------------------- >>> if (!uri==myself) { >>> route(1); >>> }; >>> >>> # ----------------------------------------------------------------- >>> if (uri==myself) { >>> >>> if (method=="REGISTER") { >>> if (!www_authorize("taipei.example.net", "subscriber")) { >>> www_challenge("taipei.example.net", "0"); >>> exit; >>> }; >>> >>> if (isflagset(5)) { >>> setflag(6); >>> }; >>> >>> save("location"); >>> exit; >>> }; >>> >>> lookup("aliases"); >>> if (!uri==myself) { >>> append_hf("P-hint: outbound alias\r\n"); >>> route(1); >>> }; >>> >>> if (!lookup("location")) { >>> sl_send_reply("404", "Not Found"); >>> exit; >>> }; >>> append_hf("P-hint: usrloc applied\r\n"); >>> if (method=="INVITE") { >>> if (uri=~"^sip:6.+@") { >>> rewritehostport("xx.xx.xx.xx:5060"); >>> xlog("L_ERR", "transfer to Asterisk"); >>> route(1); >>> exit; >>> }; >>> }; }; >>> >>> route(1); >>> } >>> >>> >>> route[1] { >>> if (subst_uri('/(sip:.*);nat=yes/\1/')){ >>> setflag(6); >>> }; >>> >>> if (isflagset(5)||isflagset(6)) { >>> route(3); >>> } >>> if (!t_relay()) { >>> sl_reply_error(); >>> }; >>> exit; >>> } >>> >>> >>> route[2]{ >>> force_rport(); >>> if (nat_uac_test("19")) { >>> if (method=="REGISTER") { >>> fix_nated_register(); >>> } else { >>> fix_nated_contact(); >>> }; >>> setflag(5); >>> }; >>> } >>> >>> >>> route[3] { >>> if (is_method("BYE|CANCEL")) { >>> unforce_rtp_proxy(); >>> } else if (is_method("INVITE")){ >>> force_rtp_proxy(); >>> t_on_failure("1"); >>> }; >>> if (isflagset(5)) >>> search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); >>> t_on_reply("1"); >>> } >>> >>> failure_route[1] { >>> if (isflagset(6) || isflagset(5)) { >>> unforce_rtp_proxy(); >>> } >>> } >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> --- >> avast! Antivirus: Inbound message clean. >> Virus Database (VPS): 0641-3, 2006/10/12 >> Tested on: 2006/10/13 ¤U¤È 05:15:28 >> avast! - copyright (c) 1988-2006 ALWIL Software. >> http://www.avast.com >> >> >> >> > >

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