- sr-users - kamailio.org

Re: [Serusers] help with error
by Klaus Darilion 03 May '07

03 May '07

A client REGISTERs and REGISTERs and REGISTERs and somehow ser does not detect that this is always the same client and creates a new entry in the location table. This may happen e.g. with buggy NATs which changes the port for every REGISTER or with buggy clients. Thus, limit the number of max entries in location table: http://www.openser.org/docs/modules/0.9.x/registrar.html#AEN167 and set desc_time_order to 1 http://www.openser.org/docs/modules/0.9.x/registrar.html#AEN167 although this is openser I think these parameters also exists in ser. There is no need to change MAX_BRANCHES. Fix the problem by limiting the entries in the location table. regards klaus Steve Blair wrote: > > Klaus: > > It seems the only way to increase the limit is to change MAX_BRANCHES > then re-compile SER. I do not think that approach is best. Can you think > of any other reason why this error would be generated? > > Thanks,Steve > > Klaus Darilion wrote: >> I guess there are more then "max-branches" entries in location table >> (for this user). >> >> regards >> klaus >> >> Steve Blair wrote: >>> >>> I'm getting the following error on a ser-0.9.7 system and I cannot >>> figure out why. I see the message originates in dset.c but the cause >>> of the error is unclear. Any thoughts? >>> >>> Thanks,Steve >>> >>> ERROR: append_branch: max nr of branches exceeded >>> lookup(): Error while appending a branch >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>

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[Serusers] shared memory query
by Gaurav Kansal 03 May '07

03 May '07

Hi The shared memory which SER process will allocate depend on value of SHM_MEM_SIZE in config.h. In a scenario where I am only establishing the calls and not releasing, sooner or later the application will run out of memory. What is the behavior of SER then? Regards, Gaurav Kansal

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[Serusers] Are there efforts to multi-thread SER?
by Heaton, Eric D 02 May '07

02 May '07

Hi All, Are there any efforts within the community here to multithread any aspect of the SER project (i.e. either in the "core" functionality, or over any of the plug-ins), for example, to have it run more efficiently, provide for more bandwidth, etc. on a multi-core CPU? I'm starting to look into the topic myself, and don't want to duplicate work; at the least, I'd want to add to it! Thanks in advance. -- Eric

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[Users] BYE message without Route header field
by rubel Ahmed 02 May '07

02 May '07

Hi, I am using Openser-1.2.0 with dialog module. When I make a call then full session is saved in dialog correctly and when bye received then the session will be free. but when the route header feild is not available in BYE message then the session is not removed at receive BYE, Because This BYE message is not caught bye dlg_onroute function. Now can I destroy this session in dlg_onreq function? Thanks BDRubel

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[Serusers] help with error
by Steve Blair 02 May '07

02 May '07

I'm getting the following error on a ser-0.9.7 system and I cannot figure out why. I see the message originates in dset.c but the cause of the error is unclear. Any thoughts? Thanks,Steve ERROR: append_branch: max nr of branches exceeded lookup(): Error while appending a branch

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[Users] Proxy/Registrar and NAT
by Alex Hermann 02 May '07

02 May '07

Hello list members, I'd like to use multiple proxies/registrars in a SRV managed failover/load-balancing situation. For NAT purposes I could use a couple of dedicated load-balancers with path support to store the NAT enabled route in the location table. The registrars have a shared location table on a database server. This situation works, but.... Is there a possibility to skip the load-balancers? I want the proxy/registrar to store the NAT-route somehow in the location table, so that when the contact needs to be contacted, the proxy can determine if it can send the request itself, or it needs to forward it via one of the other proxies (the one with which the contact has the NAT relation). I hope you understand the topology, otherwise I might need to create some ascii-art. -- Greetings, Alex Hermann Speakup

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[Serusers] Computing value of the branch parameter in SER
by Andrew Pogrebennyk 02 May '07

02 May '07

Hi, Can you tell me how SER computes the value of the branch parameter or where should I look for the algorithm? Thanks in advance! -- /"\ Best regards, | andrew.pogrebennyk (at) portaone.com \ / Andrew Pogrebennyk | http://callcc.livejournal.com/ X VoIP engineer | JID: marduk (at) jabber.lafox.net / \ ASCII Ribbon Campaign | Against HTML Mail and News

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[Serusers] iptel.org - Technical board - Suggestion - REQUEST FOR FEEDBACK
by Greger V. Teigre 02 May '07

02 May '07

I believe most people by now have had time to respond to the thread where the idea and discussion from Prague around a technical board was started (April 12). Based on the feedback received, I summarize and offer the following make-it-simple suggestion: We create an iptel.org technical board (TB: 1. The board is responsible for resolving issues that is not possible to be resolve through a community consensus process, and it is allowed to take action where it finds necessary for the benefit of the iptel.org community. 2. The board is common for all iptel.org projects. 3. The board shall consist of five (5) members and the term shall be two years. 4. A board member can stay for several terms. 5. The five members shall be selected with one representative from SEMS, two from the SER developer (cvs write access) community, and two representatives for the broader iptel.org community. 6. All seruser and serdev subscribers nominate candidates, as well as vote. 7. Electable members must be contributors to iptel.org in some way and should not be controversial among subscribers. 8. Each subscriber can give his/her vote to one or more candidates, up to the number of board members (5). Process forward: People should respond with concerns and suggestions to this post. If there is consensus for changes, I will work as the editor and write up a new suggestion. Once the community can agree to the set of rules governing the board, we will proceed to nomination of candidates and then voting. g-)

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[Users] Re: Users Digest, Vol 23, Issue 97
by wangdq＠gongye.com.cn 02 May '07

02 May '07

use rtpproxy trunk tree, perhaps have no this buf. rtpproxy-0.3.0 has this problem . I use the SVN trunk . this problem not happend. 在 2007-05-01二的 16:15 +0800，users-request(a)openser.xn--org:-7z8fk634a > Send Users mailing list submissions to > users(a)openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > > You can reach the person managing the list at > users-owner(a)openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: memory leak in presence module? (Klaus Darilion) > 2. rtpproxy CPU usage issue.. (Sajith T S) > 3. Dropped calls with OpenSER 1.1.x and Asterisk >= 1.2.14 > (Edoardo Serra) > 4. Re: Redirect the same INVITE to the SIP SERVER from Balancer? > (Alejandro Sanchez) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 30 Apr 2007 15:34:31 +0200 > From: Klaus Darilion <klaus.mailinglists(a)pernau.at> > Subject: Re: [Users] memory leak in presence module? > To: daniel(a)voice-system.ro > Cc: "users openser.org" <users(a)openser.org> > Message-ID: <4635F067.6070006(a)pernau.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi! > > I tried again and it happened again: > > Apr 30 15:00:54 ds3000 /usr/sbin/openser[7648]: > 32b24f15e52d603ba890a9729723c4b0.0167///45-6782(a)83.136.32.132 PUBLISH > detected, handle_publish ... > outside t_newtran > Apr 30 15:00:54 ds3000 /usr/sbin/openser[7655]: > 32b24f15e52d603ba890a9729723c4b0.7e11///14-6782(a)83.136.32.132 PUBLISH > detected, handle_publish ... > outside t_newtran > Apr 30 15:00:54 ds3000 /usr/sbin/openser[7648]: > 32b24f15e52d603ba890a9729723c4b0.0167///45-6782(a)83.136.32.132 PUBLISH > detected, handle_publish ... > inside t_newtran > Apr 30 15:00:54 ds3000 /usr/sbin/openser[7655]: > 32b24f15e52d603ba890a9729723c4b0.7e11///14-6782(a)83.136.32.132 PUBLISH > detected, handle_publish ... > inside t_newtran > Apr 30 15:01:03 ds3000 /usr/sbin/openser[7644]: child process 7648 > exited by a signal 9 > Apr 30 15:01:08 ds3000 /usr/sbin/openser[7644]: core was not generated > Apr 30 15:01:08 ds3000 /usr/sbin/openser[7644]: INFO: terminating due to > SIGCHLD > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7657]: INFO: signal 15 received > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7657]: Memory status (pkg): > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7657]: qm_status (0x8145960): > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7657]: heap size= 1048576 > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7659]: INFO: signal 15 received > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7653]: INFO: signal 15 received > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7653]: Memory status (pkg): > Apr 30 15:01:14 ds3000 /usr/sbin/openser[7650]: INFO: signal 15 received > > > Any hints how to debug this? > > regards > klaus > > Daniel-Constantin Mierla wrote: > > Hello Klaus, > > > > On 04/30/07 13:55, Klaus Darilion wrote: > >> > >> > >> Daniel-Constantin Mierla wrote: > >>> Hello Klaus, > >>> > >>> On 04/27/07 09:27, Klaus Darilion wrote: > >>>> Hi Daniel! > >>>> > >>>> I've tried with t_release and it was running fine several hours > >>>> without leaking. But then, unfortunately openser terminated with > >>>> signal 9. I've never seen this before. > >>> > >>> signal 9 is KILL, it is very strange if it was not issued by a user > >>> or other process. > >>> > >>> We discovered the issue (tm/uac.c, line 224), ther eis flag that is > >>> kept to see if there was some operation with the transaction, but in > >>> case of handle_publish() that flag is set by TM api when sending > >>> NOTIFY. The patch is trivial, removing a line, but we have to > >>> investigate if there are some other effects -- so it may take a > >>> while. t_release() should solve meanwhile. > >> > >> Should solve the memory-leak - but the signal 9? > > It might be that it took so long to write the mem long at shut down and > > openser killed itself. If it was not due to a openser stop, then I am > > not aware of any case when signal 9 is sent unless issued by user. > > > > Cheers, > > Daniel > > > >> > >> regards > >> klaus > >> > > > > ------------------------------ > > Message: 2 > Date: Mon, 30 Apr 2007 19:43:41 +0530 > From: Sajith T S <sajithts(a)gmail.com> > Subject: [Users] rtpproxy CPU usage issue.. > To: users(a)openser.org > Message-ID: <4635F995.3020908(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > Have any of you folks seen an abnormal CPU usage issue with rtpprpxy? > At times, "top" shows rtpproxy uses 100% CPU (sometimes 99%, sometimes > even 101%!), and it stops responding to openser. I have to restart > everything now. > > A small thread in Serdev list recommends upgrading rtpproxy to latest > in CVS: > > http://lists.iptel.org/pipermail/serdev/2006-June/007459.html > > Mine was fairly old, so I upgraded, but without much progress. What > really bugs is I don't know what triggers it. > > Backtraces suggest that this occurs in different places in rtpproxy, > before and after upgrading. > > Before upgrade: > > (gdb) where > #0 0xb7f49410 in ?? () > #1 0xbffd6d88 in ?? () > #2 0x00000076 in ?? () > #3 0xbffd2410 in ?? () > #4 0x4a8d9fd1 in sendto () from /lib/tls/libc.so.6 > #5 0x0804ac0c in main (argc=7, argv=0xbffd6e14) at main.c:1467 > > After upgrade: > > (gdb) where > #0 0xb7f0a410 in ?? () > #1 0xbfcb0258 in ?? () > #2 0xbfcadfc0 in ?? () > #3 0xbfcab8e0 in ?? () > #4 0x00bb8dd1 in recvfrom () from /lib/tls/libc.so.6 > #5 0x0804a568 in main (argc=7, argv=0xbfcb02e4) at main.c:1364 > > Any clues? Is this a known problem and is there a patch or some sort > ofworkaround? Is there anything I need to upgrade? (This is CentOS > 4.4, FWIW) Anything that should be done in openser configuration? > > Thanks much in advance. > > - Sajith. > > > > ------------------------------ > > Message: 3 > Date: Mon, 30 Apr 2007 16:27:43 +0200 > From: Edoardo Serra <edoardo.serra(a)webrainstorm.it> > Subject: [Users] Dropped calls with OpenSER 1.1.x and Asterisk >= > 1.2.14 > To: users(a)openser.org > Message-ID: <f14udl$s9t$1(a)sea.gmane.org> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > Hi guys, > I'm having a problem about asterisk dropping calls received from OpenSER > > The problems was spotted at Asterisk users mailing lists > http://lists.digium.com/pipermail/asterisk-users/2007-April/184875.html > > Asterisk complains about an ACK not received and drops the calls. > It expects OpenSER to send an ACK back to its '200 OK' when the call is > set up > > Is that a problem OpenSER should solve ??? > > Tnx in advance > > Regards > > Edoardo Serra > WeBRainstorm S.r.l. > > > > > ------------------------------ > > Message: 4 > Date: Mon, 30 Apr 2007 10:54:17 -0500 (CDT) > From: Alejandro Sanchez <alexsc_java(a)yahoo.com.mx> > Subject: Re: [Users] Redirect the same INVITE to the SIP SERVER from > Balancer? > To: daniel(a)voice-system.ro > Cc: openser openser <users(a)openser.org> > Message-ID: <626480.1349.qm(a)web35306.mail.mud.yahoo.com> > Content-Type: text/plain; charset="iso-8859-1" > > Thank's for answer Daniel. > > In fact i think is a retransmission and i need the > balanser ask to the SipServer if the another > videotelephone B is register in every retransmision of > the INVITE. > > I send the trace where: > > -192.168.23 = Balancer (openser) > -192.168.20 = SipServer (openser) > -192.168.4 = VideoTelephone > > In this case there is not another videotelefhone. > > > Thank's > > > > > > --- Daniel-Constantin Mierla <daniel(a)voice-system.ro> > escribi: > > > Hello, > > > > if it happens in very short time, then seems to be a > > retransmission. Can > > you post a sip trace (ngrep, ethereal) taken on the > > balancer? > > > > Cheers, > > Daniel > > > > > > On 04/27/07 22:01, Alejandro Sanchez wrote: > > > Hello. > > > > > > My enviroment is the next: > > > > > > -Balancer > > > OS. Red Hat 4 up. 4 > > > Openser 1.1.1 (balance with the module dispatcher) > > > > > > -Sip Server > > > OS. Red Hat 4 up. 4 > > > Openser 1.1.1 > > > > > > > > > The actual flow is this. > > > > > > VTA B C VTB > > > ---register----> > > > ---register----> > > > <---200 OK------ > > > <----200 OK----- > > > > > > ---invite------> > > > -----invite----> > > > <-404 NotFound-- > > > --404 NotFound-- > > > <----------register--------- > > > ----register----> > > > <----200 OK------ > > > ------------ 200 OK--------> > > > ----invite-----> > > > <-404 NotFound-- > > > > > > VTA= videotelephone A > > > B= balancer (openser 1.1.1 with dispatcher module) > > > C= SipServer (openser 1.1.1) > > > VTB= video telephone B > > > > > > How you see when the VTA sends again the same > > invite, > > > the balancer(B) answer with the 404 not found but > > > without ask again to the SipServer(C), and my > > problem > > > is that VTB is late and i have to try find it in > > the > > > second invite, then in the second invite i need > > ask > > > again to the SipServer(C) if the VTB is register. > > > > > > VTA y VTB access to the IP network making a > > dial-up > > > connection. > > > > > > Why the balancer dosen't send the second invite to > > > SipServer? > > > > > > How do i get this flow? > > > > > > > > > Thank's for the Help > > > > > > > > > __________________________________________________ > > > Correo Yahoo! > > > Espacio para todos tus mensajes, antivirus y > > antispam gratis! > > > Regstrate ya - http://correo.yahoo.com.mx/ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > __________________________________________________ > Correo Yahoo! > Espacio para todos tus mensajes, antivirus y antispam gratis! > Regstrate ya - http://correo.yahoo.com.mx/ >

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[Serusers] ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established
by Tuan Nguyen 01 May '07

01 May '07

Dear Gergely and everybody, When I call from UAC1 register in Auth_SER to UAC2 register in Verifer_SER i got follow error: WARNING: no fork mode 0(6759) init_tcp: using epoll_lt as the io watch method (auto detected) 0(6759) Maxfwd module- initializing 0(6759) AUTH_IDENTITY:mod_init: Private key path is missing! Authorization service is disabled 0(6759) INFO: udp_init: SO_RCVBUF is initially 107520 0(6759) INFO: udp_init: SO_RCVBUF is finally 262142 0(6759) BUG: mysql: db_init: called from the main process, ignoring... 0(6759) BUG: mysql: db_init: called from the main process, ignoring... 0(6759) BUG: mysql: db_init: called from the main process, ignoring... 0(6759) BUG: mysql: db_init: called from the main process, ignoring... 0(6759) ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established 0(6759) ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established What is T-state, how can i this problem solve? Thank for response. Tuan.

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[Serusers] verify.cfg of auth_identity module
by Tuan Nguyen 01 May '07

01 May '07

Dear Gergely, In your documentation of auth_identity, parameter of privatekey_path, certificate_path, certificate_url are belong to "Authoriter service" but in your email 24 Apr, you configure these parameter in vrfy.cfg - "Verifier service". I am confuse with this different. Can you explain this point for me? Thank for response. Tuan.

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[Users] When is RTP Proxy needed?
by lists＠infoway.net 30 Apr '07

30 Apr '07

Hi all, We are interested in developing yet-another peer-to-peer VoIP service. Something with a simple Softphone/IM-like client that allows users to talk/chat with one another. However, and pardon our technical ignorance, we don't have 100% clear if/when we need to use something like rtpproxy or mediaproxy. Basically, we want to do all the setup of the calls and keep track of CDRs. However, we're hope that the client's actual media traffic will flow directly among the users and not have to come through our servers. For practical purposes, most users will be in domestic US with some sort of xDSL service and behind NAT. Eventually, we would like to offer voicemail services and even PSTN termination (for a fee) to these users. Can anyone shed some light as to when, if really needed, will something like rtpproxy kick in? Thanks

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[Users] User Realm on Location table
by Edson 30 Apr '07

30 Apr '07

Hi... I have to manage several domains on a same SIP server, but there is a policy that prohibit/block inter-domain calls. Consider A(a)a.a, B(a)a.a and B(a)b.b. Now suppose that A(a)a.a wants to call B(a)a.a. But B(a)a.a is not connected and B(a)b.b is. After a 'lookup("location")', the RURI returned is B(a)b.b and that is not the user wanted. Location table has a column named 'Domain', but it's allways empty. How to populate this column? How to use it (make 'lookup' consider it)? And how to prevent A(a)a.a calling B(a)a.a to reach B(a)b.b? Edson

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[Users] Redirect the same INVITE to the SIP SERVER from Balancer?
by Alejandro Sanchez 30 Apr '07

30 Apr '07

Hello. My enviroment is the next: -Balancer OS. Red Hat 4 up. 4 Openser 1.1.1 (balance with the module dispatcher) -Sip Server OS. Red Hat 4 up. 4 Openser 1.1.1 The actual flow is this. VTA B C VTB ---register----> ---register----> <---200 OK------ <----200 OK----- ---invite------> -----invite----> <-404 NotFound-- --404 NotFound-- <----------register--------- ----register----> <----200 OK------ ------------ 200 OK--------> ----invite-----> <-404 NotFound-- VTA= videotelephone A B= balancer (openser 1.1.1 with dispatcher module) C= SipServer (openser 1.1.1) VTB= video telephone B How you see when the VTA sends again the same invite, the balancer(B) answer with the 404 not found but without ask again to the SipServer(C), and my problem is that VTB is late and i have to try find it in the second invite, then in the second invite i need ask again to the SipServer(C) if the VTB is register. VTA y VTB access to the IP network making a dial-up connection. Why the balancer dosen't send the second invite to SipServer? How do i get this flow? Thank's for the Help __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/

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[Serusers] How to define avp localy on he ser.cfg
by tzieleniewski 30 Apr '07

30 Apr '07

Hi, Is it possible to define avp in ser.cfg without loading it from the database?? What is the syntax? Thanks in advance! Tomasz

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[Serusers] t_relay_to_udp() method with ip address passed as avp? - ser 2_0_0
by tzieleniewski 30 Apr '07

30 Apr '07

Hi, Is it possible to pass the ip address in the function t_relay_to_udp(..) in the m module using avp. For instance: ... t_relay_to_udp("$relay_to_udp","5060"); ... When I try to do it this way I get the following error: Apr 30 14:30:03 sen /home/ser/ser/ser-2.0/ser[7552]: Maxfwd module- initializing Apr 30 14:30:03 sen ser[7552]: AVPops - initializing Apr 30 14:30:03 sen ser[7552]: ERROR: mk_proxy: could not resolve hostname: "$relay_to_udp" Apr 30 14:30:03 sen ser[7552]: ERROR: fixup_hostport2proxy: bad host name in URI <$relay_to_udp> Apr 30 14:30:03 sen ser[7552]: ERROR: fix_expr : fix_actions error Cheers Tomasz

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[Serusers] Ser2.0 wrong behaviour of the sl_reply() function?
by tzieleniewski 30 Apr '07

30 Apr '07

Hi, I am using SER rel 2_0_0 from CVS. I encountered the following situation. My scenario was the following: UA -> Application Serwer -> SER -> UA 1. UA sent an INVITE message to the application server. 2. Application Server changed the Request URI and forwarded statelessly the INVITE message to SER. 3. Everything would be ok but SER replied to the Application server instead of the client. Application server changed only the Request URI and nothing else. Shouldn't SER sent the response according to the VIA header value?? I kindly ask for Your help. Below is the ngrep: 192.168.0.112:5060 - this is application server 192.168.0.112:5061 - this is calling UA U 2007/04/25 16:17:05.652934 192.168.0.112:5060 -> 192.168.0.165:5060 INVITE sip:sen@voip.rd.touk.pl SIP/2.0. Via: SIP/2.0/UDP 192.168.0.112:5061;rport=5061;branch=z9hG4bKhnrxbtgp;received=127.0.0.1. Max-Forwards: 70. To: <sip:sen@voip.rd.touk.pl>. From: <sip:pz@voip.rd.touk.pl>;tag=jlegd. Call-ID: pfnqwekjwtuxbfv(a)192.168.0.112. CSeq: 87 INVITE. Contact: <sip:pz@192.168.0.112:5061>. Content-Type: application/sdp. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. Supported: 100rel. User-Agent: Twinkle/0.9. Content-Length: 304. . v=0. o=pz 86411392 1733597180 IN IP4 192.168.0.112. s=-. c=IN IP4 192.168.0.112. t=0 0. m=audio 8000 RTP/AVP 98 97 8 0 3 101. a=rtpmap:98 speex/16000. a=rtpmap:97 speex/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. # U 2007/04/25 16:17:05.699923 192.168.0.165:5060 -> 192.168.0.112:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.112:5061;rport=5060;branch=z9hG4bKhnrxbtgp;received=127.0.0.1;received=192.168.0.112. To: <sip:sen@voip.rd.touk.pl>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6a45. From: <sip:pz@voip.rd.touk.pl>;tag=jlegd. Call-ID: pfnqwekjwtuxbfv(a)192.168.0.112. CSeq: 87 INVITE. Proxy-Authenticate: Digest realm="voip.rd.touk.pl", nonce="462f640d527dfc442d53379e1f0167c8a12a583d", qop="auth". Server: Sip EXpress router (2.0.0-rc1 (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.0.165:5060 "Noisy feedback tells: pid=20441 req_src_ip=192.168.0.112 req_src_port=5060 in_uri=sip:sen@voip.rd.touk.pl out_uri=sip:sen@voip.rd.touk.pl via_cnt==1". . Cheers Tomasz

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[Serusers] load_attrs("$tu","@ruri") not working - SER 2_0_0
by tzieleniewski 30 Apr '07

30 Apr '07

Hi, I was trying to load user attributes form the database (which where previously defined and set through serweb) and after I tried to load them with load_attrs("$tu","@ruri") no attributes appeared for the user identified by ruri. When I loaded attributes using load_attrs("$tr","@ruri") from the uri_attrs table then everything was ok and attributes were loaded. Please point me what do I missed so the attributes are not loaded? Below is my ser.cfg code and database contents: if (lookup_user("$tu.uid", "@ruri")) { xlog("L_INFO", " route[INBOUND]: load_attrs($tu,@ruri) <%rm> <%ru> \n"); load_attrs("$tu", "@ruri"); dump_attrs(); xlog("L_INFO", " route[INBOUND]: load_attrs($tr,@ruri) <%rm> <%ru> \n"); load_attrs("$tr", "@ruri"); dump_attrs(); .... } database: ser@sen:~/ser/ser-2.0/serctl$ ./ser_db show user_attrs | grep testser 32 2 testser(a)voip.rd.touk.pl testser first_name 32 2 testser(a)voip.rd.touk.pl last_name 33 2 testser(a)voip.rd.touk.pl tzl(a)touk.pl email 32 2 testser(a)voip.rd.touk.pl phone 32 2 testser(a)voip.rd.touk.pl Europe/Warsaw timezone 33 2 testser(a)voip.rd.touk.pl 2007-02-26 13:50:18 datetime_created 32 0 testser(a)voip.rd.touk.pl 0 sw_allow_find 32 0 testser(a)voip.rd.touk.pl 0 sw_send_missed 33 2 testser(a)voip.rd.touk.pl sip:touk2@voip.touk.pl forward_blind 33 2 testser fd for and output in the log file: Apr 30 11:11:40 sen ser[29958]: route[INBOUND]: load_attrs($tu,@ruri) <INVITE> <sip:testser@voip.rd.touk.pl> Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:540: class=GLOBAL Apr 30 11:11:40 sen ser[29958]: AVP["lang"]="en" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:550: track=FROM class=DOMAIN Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:560: track=TO class=DOMAIN Apr 30 11:11:40 sen ser[29958]: AVP["did"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: AVP["digest_realm"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:570: track=FROM class=USER Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:580: track=TO class=USER Apr 30 11:11:40 sen ser[29958]: AVP["uid"]="testser(a)voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:590: track=FROM class=URI Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:600: track=TO class=URI Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: route[INBOUND]: load_attrs($tr,@ruri) <INVITE> <sip:testser@voip.rd.touk.pl> Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:540: class=GLOBAL Apr 30 11:11:40 sen ser[29958]: AVP["lang"]="en" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:550: track=FROM class=DOMAIN Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:560: track=TO class=DOMAIN Apr 30 11:11:40 sen ser[29958]: AVP["did"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: AVP["digest_realm"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:570: track=FROM class=USER Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:580: track=TO class=USER Apr 30 11:11:40 sen ser[29958]: AVP["uid"]="testser(a)voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:590: track=FROM class=URI Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:600: track=TO class=URI Apr 30 11:11:40 sen ser[29958]: AVP["forward_blindd"]="sip:hellboy@voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: AVP["forward_voicemaill"]="sip:hellboy@voip.rd.touk.pl" Cheers Tomasz

2 1

[Users] who removes top most route header? (openser 1.2.0 + wesip)
by Jose Antonio Garvayo 30 Apr '07

30 Apr '07

Hi, I have testing openser 1.2.0 with seas module and wesip. If I send an INVITE message with a Route header, the SIP servlet receives the INVITE with the full Route Header, and when I proxy it the Route is the same, but I expected that the top most Route header has been removed. Who is in charge to do it? openser or wesip? I first thought it might be openser through loose_route(), but it seems not to be the case. The openser cfg file is similar to the example given in wesip tutorial: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; xlog("L_INFO", "$rm: ci=$ci ft=$ft tt=$tt cseq=$cs ruri:$ru received from: $si:$sp \n"); if (!method=="REGISTER") record_route(); if (loose_route()) { xlog("L_INFO", "RR enforced\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { xlog("L_INFO", "uri == myself"); if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { xlog("L_INFO", "outbound alias\n"); append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; xlog("L_INFO", "usrloc applied\n"); append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if(!as_relay_t("app_server_one")){ t_reply("500","Application Server error"); } } route[2] { xlog ("L_INFO", "$rm: cseq:$cs $ru: RELAYING (PROXY) to $du (flags: $mF) ...\n"); if (!t_relay()) { sl_reply_error(); }; exit; } This the log from wesip: 14:43:37 27abr2007 DEBUG SeasMessageEvent [SeasRouter]- New message received, from 10.161.14.10:6025 to 10.161.14.10:7000 over UDP 14:43:37 27abr2007 DEBUG SeasTransactionEvent [SeasRouter]- hash_index = 60870,label=1842040723 flags=0 14:43:37 27abr2007 DEBUG ExpressMessageChannel [ExpressMChannel[76]]- peerAddress = 10.161.14.10/6025 14:43:37 27abr2007 DEBUG ExpressMessageChannel [ExpressMChannel[76]]- About to process INVITE sip:user2@mot.net SIP/2.0 14:43:37 27abr2007 DEBUG SipConnector [ExpressMChannel[76]]- createProcessor: Reusing existing processor 14:43:37 27abr2007 INFO SipProcessor [ExpressMChannel[76]]- An incoming message is being assigned 14:43:37 27abr2007 DEBUG ExpressMessageChannel [ExpressMChannel[76]]- Done processing INVITE sip:user2@mot.net SIP/2.0 14:43:37 27abr2007 DEBUG SipProcessor [SipProcessor[3]]- <<<<<<<<< Request Received <<<<<<<<< INVITE sip:user2@mot.net SIP/2.0 Via: SIP/2.0/UDP 10.161.14.10:6025;branch=z9hG4bK1-13972359389@10.161.14.101 Contact: <sip:10.161.14.10:6025;transport=UDP> Max-Forwards: 14 From: <sip:user1@mot.net>;tag=6661 To: <sip:user2@mot.net> Supported: 100rel Route: <sip:10.161.14.10:7000;lr=on>,<sip:10.161.14.10:6025;lr=on> Call-ID: 1-13972359389(a)10.161.14.10 CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,UPDATE,MESSAGE,PRACK,REFER Content-Type: application/sdp Content-Length: 337 14:43:37 27abr2007 DEBUG SeasHashTable [SipProcessor[3]]- Inserting hashIdx:60870 label:1842040723 into hashTable 14:43:37 27abr2007 DEBUG SipProcessor [SipProcessor[3]]- Transaction [ z9hG4bK1-13972359389(a)10.161.14.101] is new Transaction?? --> true 14:43:37 27abr2007 DEBUG EngineSipMapper [SipProcessor[3]]- Request from 10.161.14.10 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] 14:43:37 27abr2007 DEBUG ConvergedEngineValve [SipProcessor[3]]- Invoking Host:localhost 14:43:37 27abr2007 DEBUG ConvergedHostValve [SipProcessor[3]]- SipSession not found for requestINVITE, trying match against Rules 14:43:37 27abr2007 DEBUG ConvergedHost [SipProcessor[3]]- Mapped request to:/sampleservlet 14:43:37 27abr2007 DEBUG ConvergedContextValve [SipProcessor[3]]- new session created :1-13972359389@10.161.14.10 14:43:37 27abr2007 INFO ExpressMessageChannel [SipProcessor[3]]- Creating Express message channel 10.161.14.10/7000 14:43:37 27abr2007 DEBUG ExpressMessageChannel [SipProcessor[3]]- setting spiral=false 14:43:37 27abr2007 DEBUG SipRequest [SipProcessor[3]]- >>>>>>>>> Sending Request >>>>>>>>> INVITE sip:user2@mot.net SIP/2.0 Max-Forwards: 14 From: <sip:user1@mot.net>;tag=83FBA2EBC5A9EE4615D194DECA1EE955 To: <sip:user2@mot.net> Supported: 100rel Route: <sip:10.161.14.10:7000;lr=on>,<sip:10.161.14.10:6025;lr=on> CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,UPDATE,MESSAGE,PRACK,REFER Content-Type: application/sdp Call-ID: 1-13972359389(a)10.161.14.10 Contact: <sip:10.161.14.10:7000;transport=udp> P-AS-Hint: B2BUA Via: SIP/2.0/UDP 10.161.14.10:7000;branch=z9hG4bK2137041384 Content-Length: 337 Thanks in advance

2 3

[Serusers] table in database ser
by Tuan Nguyen 30 Apr '07

30 Apr '07

Hi, I have created ser database for run ser. but I got follows error: 0(8054) init_tcp: using epoll_lt as the io watch method (auto detected) 0(8054) Maxfwd module- initializing 0(8054) submit_query: Table 'ser.global_attrs' doesn't exist 0(8054) db_query: Error while submitting query 0(8054) gflags:load_attrs: Error while quering database 0(8054) init_mod(): Error while initializing module gflags ERROR: error while initializing modules Can someone tell me where talk about which table i have to create in database ser. I can't find it. Many thanks for response! Tuan.

1 0

[Users] WeSIP redundancy/failover
by Andreas Granig 30 Apr '07

30 Apr '07

Hi, I was wondering if there's any mechanism in WeSIP for redundancy and failover. Assuming that I've one application server holding SIP dialog states, how could I add high availability to it? There's some sort of session replication in Tomcat [1], so is this somehow used by WeSIP? Are there any other best practices to redundantly set up WeSIP? Andreas [1] http://www.onjava.com/pub/a/onjava/2004/11/24/replication1.html This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

2 1

[Users] Forward Sessions to another SIP server
by konstantinos nikou 30 Apr '07

30 Apr '07

Dear all, I have the following issue: I would like my calls coming to openSER with URI 40XXXXXXX(a)abc.com to be forwarded to another SIP server with IP e.g. 10.10.0.1 with URI +140XXXXXXX(a)domain.com. Could someone tell me how i could do it? Thank you very much. Best regards, K.N. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] Problem with avp_db_load in OpenSER 1.2
by Daryl Sanders 29 Apr '07

29 Apr '07

I cannot seem to get my usr_preferences to load using the following load statement. avp_db_load("$avp(s:caller_username)", "*"); I have verified with xlog and avp_print() that $avp(s:caller_username) does contain a valid value before calling avp_db_load, however nothing is returned. It will work if I use this: avp_db_load("$fu/username", "*"); I really need to make it work via $avp(s:caller_username) if possible because my call forwarding and custom call routing features change this value often. Thanks, - Daryl

2 1

[Users] Openser Summit 2007
by Flavio Goncalves 29 Apr '07

29 Apr '07

Hello, Is there any plans for an OpenSER summit 2007? Flavio.

2 1

[Serusers] problem with MySQL
by Tuan Nguyen 29 Apr '07

29 Apr '07

Hi everyone, + I have problem with mysql when run ser: WARNING: no fork mode and more than one listen address found (will use only the the first one) 0(12395) init_tcp: using epoll_lt as the io watch method (auto detected) 0(12395) Maxfwd module- initializing 0(12395) new_connection: Access denied for user 'root'@'localhost' (using password: YES) 0(12395) gflags:mod_init: Error while connecting database 0(12395) init_mod(): Error while initializing module gflags ERROR: error while initializing modules in shell, when i run mysql -u root -pxxxxx, it works. but when run ser as root, it don't work, i tried to fix this problem 2 days but can't. can someone help me and tell me where is problem. + second question: in serusers archive don't have search box, when I want to search someone has the same problem like me and know how they solve it, i have to look every threads or we have other way to search efficient, I tried to search with google group but can't. thank you very much for your response. cheers Tuan.

1 0

[Users] new CDRTool release 5.1.1
by Adrian Georgescu 29 Apr '07

29 Apr '07

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog from 5.0.10: cdrtool (5.1.1) unstable; urgency=low * Updated sample data for rating tables * Updated mysql schema * Fixed bug in csv import script cdrtool (5.1.0) unstable; urgency=low * Added historical rating engine, requires changes in database structure (you must apply changes from setup/mysql/ alter_tables.mysql) * Must add in global.inc "AccountsDBClass" => "DB_ser" for each SER data-source, there is no default anymore * Added function to obfuscate caller Ids, this changes the login rights The login accounts rights must be updated in the Accounts page * Added capability to do accounting for SIP URIs (e.g. paid helpdesk) * Improved compatibility with Firefox in media sessions screen * Added rights for login accounts to show pricing and caller id * Removed unused NGN rating (billing_customers table has been modified, please update your CSV file import tools) * Modify index of rates tables to accomodate multiple applications (you must apply changes from setup/mysql/alter_tables.mysql) * Default application set to 'audio', application field is now mandatory when importing rates CSV files * Use separate name spaces for memcache keys per data source * Allow quota checks for multiple OpenSER data-sources * Show memory usage in syslog * Print sql errors in syslog Regards, Adrian Georgescu

1 0

[Users] A problem wiht openseradmin0.3--Add New Account Preference
by Lee Jacky 29 Apr '07

29 Apr '07

Hi,everyone,I am jacky.I use openser1.1+openseradmin0.3. Now I am troubled with a problem,and here is the detail: After I install the openseradmin, I add attribute Tel to the account_preferences_type,and then I click the link:Add New Account Preference. I fill the form like this: username:admin domain:voip.net attribute:Tel values:12456 Then I click the button:Create, and it redirect to a error page,whose content is as follows: ArgumentError in Voip account preferencesController#create wrong number of arguments (1 for 2) RAILS_ROOT: /usr/local/openseradmin-0.3/script/../config/.. Application Trace | Framework Trace | Full Trace /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1700:in `attribute=' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1700:in `send' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1700:in `attributes=' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1699:in `each' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1699:in `attributes=' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1532:in `initialize_without_callbacks' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/callbacks.rb:225:in `initialize' #{RAILS_ROOT}/app/controllers/voip_account_preferences_controller.rb:27:in `new' #{RAILS_ROOT}/app/controllers/voip_account_preferences_controller.rb:27:in `create' Request Parameters: {"voip_account_preference"=>{"username"=>"admin", "domain"=>"voip.net", "value"=>"12456", "attribute"=>"Tel"}, "commit"=>"Create"} I have considerde it for a long time ,but I still can't solve it, so I hope you can help me to give me some advice, Any suggestion will be appreciated. Best wishes, Yours, Jacky ___________________________________________________________ 雅虎免费邮箱3.5G容量，20M附件！ http://cn.mail.yahoo.com/

1 0

[Users] WeSIP with openser 1.2.0 again
by tele 28 Apr '07

28 Apr '07

Hi, fresh install of OpenSER 1.2.0 and wesip-05_03_07-beta-0.1.3.tar.gz. i'm able to run wesip and successful deploy my app, but when the first INVITE come into wesip i get: java.lang.StringIndexOutOfBoundsException: String index out of range: -37 at java.lang.String.checkBounds(String.java:398) at java.lang.String.<init>(String.java:443) at com.voztele.javax.sip.header.ExtensionHeaderImpl.<init>(ExtensionHeaderImpl.java:40) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:243) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:619) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:619) I'm using JDK/JRE version 6. With old version of wesip and openser i've no problem. thanks :tele

3 5

RE: [Users] WeSIP setup problem
by ventilo35＠free.fr 28 Apr '07

28 Apr '07

Hi Gines, thanks for your support > Your servlet named "neo" is being executed and looks like it is > proxying but what from what I see in the packet it is proxying to > itself. Could you try to proxy to another address and see if this > still happens ? Aside, could you send me your configuration files and > your servlet code ? The serlvet code is the following protected void doInvite(SipServletRequest callerInvite) throws ServletException, IOException { //Get a proxy for this request Proxy callerProxy = callerInvite.getProxy(); System.out.println("doInvite within neos servlet: requestURI is " + callerInvite.getRequestURI() ); callerProxy.proxyTo(callerInvite.getRequestURI()); } This code causes the "no route" exception. The println above gives : doInvite within neos servlet: requestURI is sip:5002@192.168.1.11 where 5002 is the called SIP phone and 192.168.1.11 is the address of my OpenSER + WeSIP box (i.e. the managed SIP domain) Following your advice I have tried to change the proxyTo to a different address: SipURI dest = sipFactory.createSipURI("5002","192.168.1.16"); callerProxy.proxyTo(dest); It that case, the called phone (5002, IP address 192.168.1.16) gets the INVITE correctly. However, something puzzles me here: the serlvet is not supposed to know the IP address of the called phone. In my understanding, the AS processes the INVITE, then it must send it back to the proxy server which is supposed to forward it to the next hop, is that right ? Could you please give me an example of a doInvite which basically does nothing and which lets the INVITE be forwarded to the called party ? Finally here are my config files: <<<<<<<<< Server.xml >>>>>>>>> <Server port="5005" shutdown="SHUTDOWN" debug="1"> <Service name="NEOS-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="7080" minProcessors="5" maxProcessors="5" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="5" enableLookups="true" acceptCount="10" debug="1" addresses="192.168.1.11:5060" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="127.0.0.1" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5080" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="javax.sip.STACK_NAME" value="neos_as" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="10" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="10" /> <Property key="com.voztele.javax.sip.SPIRAL_HDR" value="true" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="3" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="localhost" debug="1" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="localhost" nameSip="192.168.1.11_5060" debug="1" appBase="wesipapps" className="com.voztele.sipservlet.core.ConvergedHost"> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/> </Host> </Engine> </Service> </Server> <<<<<<<<<<openser.cfg>>>>>>>>>>>>>> # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes # listen = tcp:192.168.1.11:5060 listen = udp:192.168.1.11:5060 # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # added for WeSIP loadmodule "seas.so" modparam("seas","listen_sockets","127.0.0.1:5080") # end added for WeSIP # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # ftoutain essai de supprimer Ã§a ? ne change rien # 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"); 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] { if (is_present_hf("X-WeSIP-SPIRAL")) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (! as_relay_t("neos_as") ){ t_reply("500", "NEOS service not found"); }; exit; }

3 2

[Users] Re: Multiple invites
by raviprakash sunkara 28 Apr '07

28 Apr '07

Sorry Venkatesh an U me ..... to my mobile Am Not in hyd , I left My Company, In the mail list if U see the Klaus mail , he raply and U can do , if U knew the Avpops modules and pesudo attributes, http://www.openser.org/dokuwiki/doku.php/tutorials:avpops#serial_forking see that url , is the serial_forking , Do in the Same in parallel _ forking other call me , before on 2nd of the May. On 4/25/07, venkatesh.d <venkatesh.d(a)pyronetworks.com> wrote: > > hai raviprakash, > > i have tried for that multiple invites behind NAT and > Mediaproxy and calling from A(a)sipdomain.com to B(a)sipdomain.com and same > call have to forword to C(a)sipdomain.com . But C(a)sipdomain.com > invite message going to B system . > if you feel free can you send your openser.cfg file NATING > with out using media proxy and with multiple invites > > > Thanks and Regards > venkatesh.d > -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

[Users] Openser with Quintum voice gateway and an important load
by mailinglist-openser-users 27 Apr '07

27 Apr '07

Hi, Openser version: 1.2.0 We have 4 central Quintum CMS platform hosted in our computer room. We have several tens of Qintum voice gateway ( DX / AX / Call Relay ) on remote customer site. Our customer are mainly call centers so they generate a lot of small calls with an important number of call wich are never established ( around 50% ). The central platform is quite loaded, with around 600 simultaneous calls and around 20 new call per second. Up to now, we used H323 as the signalisation protocol. Now we are planning to migrate to SIP and use openser 1.2.0 for that. Does anybody already use openser with Quintum as voice gateway with so many calls ? I'd like to share experience with people using this kind of setup with this kind of load. There are no know problem between openser and Quintum SIP implementation ( we use firmware P104-12-16 ) ? Openser should be able to route all these call towards our 4 central gateway with LCR while as well sending radius accounting. Our main and backup SIP server are IBM X330 ( Bi-Pentium III 1266 Mhz with 2GB of memory ). Will it be enough to handle 600 simultaneous call and 20 new calls per seconds ? For radius accounting, we have 2 radius servers. It seems that when the first adius server is down, openser is trying 3 times the first radius server and then go to the second one but the children seems to be blocked during this time ( for example in debug mode, no more new call can be establised while radius accounting is not yet finished ). So should I start lots of children ( 20 or more ) so that if the first radius is down, openser will still be able to route new calls while some children are blocked for accounting reason ? As well I'm planning to use the "dialog" module, if this module largely increasing the load on openser ? Regards, Jerome. -- Jerome

2 1

[Users] How can I pass pseudovariables to functions?
by Peter Lemenkov 27 Apr '07

27 Apr '07

Hello All! I have a question. If I declare variable like this: $var(ip_address) = "192.168.0.1"; I can't use this variable in functions like that one: revritehostport ($var(ip_address)); It leads me to the following message: omega1 ~ # /usr/local/sbin/openser -f /usr/local/etc/openser/openser_registrar.cfg -l 192.168.0.1 0(5723) parse error (316,25-37): syntax error 0(5723) parse error (316,37-38): bad argument, string expected ERROR: bad config file (2 errors) So the question is %subj%. -- With best regards!

2 1

[Users] SER failure
by Adeyeye Michael 27 Apr '07

27 Apr '07

Pls find attached my wireshark. I tried communicating between two clients which were operating at diff. ports but futile. I likewise have no information about it on the wireshark but my SER is running. what could be wrong? MIke. No. Time Source Destination Protocol Info 1 0.000000 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 1 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 2 0.046975 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 2 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 3 0.206926 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 3 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 4 0.321780 10.128.1.115 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 4 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: KatronCo_cc:fb:e5 (00:40:f6:cc:fb:e5), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.115 (10.128.1.115), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 5 0.549158 10.128.1.34 255.255.255.255 DHCP DHCP Request - Transaction ID 0xdb812c1a Frame 5 (342 bytes on wire, 342 bytes captured) Ethernet II, Src: LucentTe_1f:33:11 (00:60:1d:1f:33:11), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.34 (10.128.1.34), Dst: 255.255.255.255 (255.255.255.255) User Datagram Protocol, Src Port: bootpc (68), Dst Port: bootps (67) Bootstrap Protocol No. Time Source Destination Protocol Info 6 0.553028 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 6 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 7 0.749929 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 7 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 8 1.072845 10.128.1.115 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 8 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: KatronCo_cc:fb:e5 (00:40:f6:cc:fb:e5), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.115 (10.128.1.115), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 9 1.499894 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 9 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 10 1.552328 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 10 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 11 1.594316 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 11 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 12 1.760680 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 12 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 13 1.823846 10.128.1.115 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 13 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: KatronCo_cc:fb:e5 (00:40:f6:cc:fb:e5), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.115 (10.128.1.115), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 14 2.552149 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 14 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 15 3.148288 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 15 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 16 3.310215 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 16 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 17 4.665102 Cisco_48:0c:92 AirvastT_0f:a2:f8 FC [Malformed Packet] Frame 17 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Cisco_48:0c:92 (00:11:20:48:0c:92), Dst: AirvastT_0f:a2:f8 (00:0a:e9:0f:a2:f8) MDS Header(Unknown(0)/Unknown(0)) [Malformed Packet: FC] No. Time Source Destination Protocol Info 18 4.697678 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 18 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 19 4.859535 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 19 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 20 6.247176 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 20 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 21 6.396495 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 21 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 22 7.784134 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 22 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 23 7.943901 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 23 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 24 9.331442 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 24 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 25 9.491392 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 25 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 26 10.878769 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 26 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 27 11.038738 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 27 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 28 12.283224 00000000.0007e9e707e0 00000000.ffffffffffff IPX RIP Response Frame 28 (60 bytes on wire, 60 bytes captured) IEEE 802.3 Ethernet Logical-Link Control Internetwork Packet eXchange IPX Routing Information Protocol No. Time Source Destination Protocol Info 29 12.426180 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 29 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 30 12.585937 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 30 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 31 13.973466 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 31 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 32 14.133145 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 32 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 33 15.520681 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 33 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 34 15.658463 Cisco_48:0c:92 AirvastT_0f:a2:f8 FC [Malformed Packet] Frame 34 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Cisco_48:0c:92 (00:11:20:48:0c:92), Dst: AirvastT_0f:a2:f8 (00:0a:e9:0f:a2:f8) MDS Header(Unknown(0)/Unknown(0)) [Malformed Packet: FC] No. Time Source Destination Protocol Info 35 15.686439 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 35 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 36 16.550969 10.128.1.34 255.255.255.255 DHCP DHCP Request - Transaction ID 0xdb812c1a Frame 36 (342 bytes on wire, 342 bytes captured) Ethernet II, Src: LucentTe_1f:33:11 (00:60:1d:1f:33:11), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.34 (10.128.1.34), Dst: 255.255.255.255 (255.255.255.255) User Datagram Protocol, Src Port: bootpc (68), Dst Port: bootps (67) Bootstrap Protocol No. Time Source Destination Protocol Info 37 16.559014 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 37 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 38 17.074193 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 38 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 39 17.237098 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 39 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 40 17.549480 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 40 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 41 17.646138 137.158.152.199 10.128.1.52 TCP [TCP segment of a reassembled PDU] Frame 41 (348 bytes on wire, 348 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39099 (39099), Seq: 0, Ack: 0, Len: 282 No. Time Source Destination Protocol Info 42 17.646163 10.128.1.52 137.158.152.199 TCP 39099 > webcache [ACK] Seq=0 Ack=282 Win=1728 Len=0 TSV=59340598 TSER=42207662 Frame 42 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39099 (39099), Dst Port: webcache (8080), Seq: 0, Ack: 282, Len: 0 No. Time Source Destination Protocol Info 43 17.646167 137.158.152.199 10.128.1.52 HTTP HTTP/1.0 200 OK (text/plain) Frame 43 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39099 (39099), Seq: 282, Ack: 0, Len: 0 Hypertext Transfer Protocol Line-based text data: text/plain No. Time Source Destination Protocol Info 44 17.646334 10.128.1.52 137.158.152.199 TCP 39099 > webcache [FIN, ACK] Seq=0 Ack=283 Win=1728 Len=0 TSV=59340598 TSER=42207662 Frame 44 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39099 (39099), Dst Port: webcache (8080), Seq: 0, Ack: 283, Len: 0 No. Time Source Destination Protocol Info 45 17.647909 137.158.152.199 10.128.1.52 TCP webcache > 39099 [ACK] Seq=283 Ack=1 Win=33303 Len=0 TSV=42207662 TSER=59340598 Frame 45 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39099 (39099), Seq: 283, Ack: 1, Len: 0 No. Time Source Destination Protocol Info 46 17.677994 10.128.1.52 137.158.152.199 TCP 39100 > webcache [SYN] Seq=0 Len=0 MSS=1460 TSV=59340606 TSER=0 WS=2 Frame 46 (74 bytes on wire, 74 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 0, Len: 0 No. Time Source Destination Protocol Info 47 17.679170 137.158.152.199 10.128.1.52 TCP webcache > 39100 [SYN, ACK] Seq=0 Ack=1 Win=131070 Len=0 MSS=1460 WS=1 TSV=42207665 TSER=59340606 Frame 47 (78 bytes on wire, 78 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39100 (39100), Seq: 0, Ack: 1, Len: 0 No. Time Source Destination Protocol Info 48 17.679188 10.128.1.52 137.158.152.199 TCP 39100 > webcache [ACK] Seq=1 Ack=1 Win=5840 Len=0 TSV=59340606 TSER=42207665 Frame 48 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 1, Ack: 1, Len: 0 No. Time Source Destination Protocol Info 49 17.679241 10.128.1.52 137.158.152.199 TCP [TCP segment of a reassembled PDU] Frame 49 (703 bytes on wire, 703 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 1, Ack: 1, Len: 637 No. Time Source Destination Protocol Info 50 17.947952 10.128.1.52 137.158.152.199 TCP [TCP Retransmission] [TCP segment of a reassembled PDU] Frame 50 (703 bytes on wire, 703 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 1, Ack: 1, Len: 637 No. Time Source Destination Protocol Info 51 18.041756 137.158.152.199 10.128.1.52 TCP webcache > 39100 [ACK] Seq=1 Ack=638 Win=66608 Len=0 TSV=42207702 TSER=59340674 Frame 51 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39100 (39100), Seq: 1, Ack: 638, Len: 0 No. Time Source Destination Protocol Info 52 18.041778 10.128.1.52 137.158.152.199 HTTP POST http://wwwl.meebo.com/mcmd/events HTTP/1.1 (application/x-www-form-urlencoded) Frame 52 (154 bytes on wire, 154 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 638, Ack: 1, Len: 88 [Reassembled TCP Segments (725 bytes): #50(637), #52(88)] Hypertext Transfer Protocol Line-based text data: application/x-www-form-urlencoded No. Time Source Destination Protocol Info 53 18.141936 137.158.152.199 10.128.1.52 TCP webcache > 39100 [ACK] Seq=1 Ack=726 Win=66608 Len=0 TSV=42207712 TSER=59340697 Frame 53 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39100 (39100), Seq: 1, Ack: 726, Len: 0 No. Time Source Destination Protocol Info 54 18.224398 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 54 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 55 18.549301 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 55 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 56 18.624740 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 56 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 57 18.784602 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 57 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 58 18.974370 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 58 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 59 19.491339 10.128.1.130 10.128.255.255 BROWSER Host Announcement COEOPNET, Workstation, Server, NT Workstation, Potential Browser Frame 59 (243 bytes on wire, 243 bytes captured) Ethernet II, Src: Micro-St_38:cf:d8 (00:13:d3:38:cf:d8), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.130 (10.128.1.130), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-dgm (138), Dst Port: netbios-dgm (138) NetBIOS Datagram Service SMB (Server Message Block Protocol) SMB MailSlot Protocol Microsoft Windows Browser Protocol No. Time Source Destination Protocol Info 60 19.724326 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 60 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 61 20.172354 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 61 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 62 20.322067 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 62 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 63 21.709846 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 63 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 64 21.869892 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 64 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 65 23.097494 10.128.1.41 10.128.255.255 BROWSER Host Announcement OLABISI, Workstation, Server, Print Queue Server, NT Workstation, Potential Browser Frame 65 (247 bytes on wire, 247 bytes captured) Ethernet II, Src: Intel_0e:98:95 (00:11:11:0e:98:95), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.41 (10.128.1.41), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-dgm (138), Dst Port: netbios-dgm (138) NetBIOS Datagram Service SMB (Server Message Block Protocol) SMB MailSlot Protocol Microsoft Windows Browser Protocol No. Time Source Destination Protocol Info 66 23.257370 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 66 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 67 23.417014 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 67 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP)

2 1

Re: [Users] NAT ping best methods
by Jayesh Nambiar 27 Apr '07

27 Apr '07

Hi Jerome, Thanks for the quick reply. I had used Openser to send NAT ping to all the registered UAs, but how do you send an OPTIONS message from Openser to the UAs registered in the location table. Also does the UAs reply as 200OK to the OPTIONS message or is it only one way. Can you please explain a little more. Thanks a lot in advance. w/regards, Jayesh ----- Original Message ---- From: Jerome Martin <jmartin(a)longphone.fr> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Friday, 27 April, 2007 6:13:31 PM Subject: Re: [Users] NAT ping best methods Hi, We are working with natted PAP2, and openser is sending keepalive (SIP OPTIONS). This is working without a glitch, with various types of CPE NAT routers. Cheers, Jerome Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/

2 1

[Serusers] SER failure?
by Adeyeye Michael 27 Apr '07

27 Apr '07

Pls find attached my wireshark. I tried communicating between two clients which were operating at diff. ports but futile. I likewise have no information about it on the wireshark but my SER is running. what could be wrong? MIke No. Time Source Destination Protocol Info 1 0.000000 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 1 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 2 0.046975 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 2 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 3 0.206926 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 3 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 4 0.321780 10.128.1.115 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 4 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: KatronCo_cc:fb:e5 (00:40:f6:cc:fb:e5), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.115 (10.128.1.115), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 5 0.549158 10.128.1.34 255.255.255.255 DHCP DHCP Request - Transaction ID 0xdb812c1a Frame 5 (342 bytes on wire, 342 bytes captured) Ethernet II, Src: LucentTe_1f:33:11 (00:60:1d:1f:33:11), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.34 (10.128.1.34), Dst: 255.255.255.255 (255.255.255.255) User Datagram Protocol, Src Port: bootpc (68), Dst Port: bootps (67) Bootstrap Protocol No. Time Source Destination Protocol Info 6 0.553028 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 6 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 7 0.749929 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 7 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 8 1.072845 10.128.1.115 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 8 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: KatronCo_cc:fb:e5 (00:40:f6:cc:fb:e5), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.115 (10.128.1.115), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 9 1.499894 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 9 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 10 1.552328 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 10 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 11 1.594316 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 11 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 12 1.760680 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 12 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 13 1.823846 10.128.1.115 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 13 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: KatronCo_cc:fb:e5 (00:40:f6:cc:fb:e5), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.115 (10.128.1.115), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 14 2.552149 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 14 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 15 3.148288 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 15 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 16 3.310215 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 16 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 17 4.665102 Cisco_48:0c:92 AirvastT_0f:a2:f8 FC [Malformed Packet] Frame 17 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Cisco_48:0c:92 (00:11:20:48:0c:92), Dst: AirvastT_0f:a2:f8 (00:0a:e9:0f:a2:f8) MDS Header(Unknown(0)/Unknown(0)) [Malformed Packet: FC] No. Time Source Destination Protocol Info 18 4.697678 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 18 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 19 4.859535 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 19 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 20 6.247176 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 20 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 21 6.396495 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 21 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 22 7.784134 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 22 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 23 7.943901 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 23 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 24 9.331442 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 24 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 25 9.491392 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 25 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 26 10.878769 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 26 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 27 11.038738 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 27 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 28 12.283224 00000000.0007e9e707e0 00000000.ffffffffffff IPX RIP Response Frame 28 (60 bytes on wire, 60 bytes captured) IEEE 802.3 Ethernet Logical-Link Control Internetwork Packet eXchange IPX Routing Information Protocol No. Time Source Destination Protocol Info 29 12.426180 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 29 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 30 12.585937 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 30 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 31 13.973466 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 31 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 32 14.133145 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 32 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 33 15.520681 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 33 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 34 15.658463 Cisco_48:0c:92 AirvastT_0f:a2:f8 FC [Malformed Packet] Frame 34 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Cisco_48:0c:92 (00:11:20:48:0c:92), Dst: AirvastT_0f:a2:f8 (00:0a:e9:0f:a2:f8) MDS Header(Unknown(0)/Unknown(0)) [Malformed Packet: FC] No. Time Source Destination Protocol Info 35 15.686439 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 35 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 36 16.550969 10.128.1.34 255.255.255.255 DHCP DHCP Request - Transaction ID 0xdb812c1a Frame 36 (342 bytes on wire, 342 bytes captured) Ethernet II, Src: LucentTe_1f:33:11 (00:60:1d:1f:33:11), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.34 (10.128.1.34), Dst: 255.255.255.255 (255.255.255.255) User Datagram Protocol, Src Port: bootpc (68), Dst Port: bootps (67) Bootstrap Protocol No. Time Source Destination Protocol Info 37 16.559014 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 37 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 38 17.074193 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 38 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 39 17.237098 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 39 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 40 17.549480 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 40 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 41 17.646138 137.158.152.199 10.128.1.52 TCP [TCP segment of a reassembled PDU] Frame 41 (348 bytes on wire, 348 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39099 (39099), Seq: 0, Ack: 0, Len: 282 No. Time Source Destination Protocol Info 42 17.646163 10.128.1.52 137.158.152.199 TCP 39099 > webcache [ACK] Seq=0 Ack=282 Win=1728 Len=0 TSV=59340598 TSER=42207662 Frame 42 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39099 (39099), Dst Port: webcache (8080), Seq: 0, Ack: 282, Len: 0 No. Time Source Destination Protocol Info 43 17.646167 137.158.152.199 10.128.1.52 HTTP HTTP/1.0 200 OK (text/plain) Frame 43 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39099 (39099), Seq: 282, Ack: 0, Len: 0 Hypertext Transfer Protocol Line-based text data: text/plain No. Time Source Destination Protocol Info 44 17.646334 10.128.1.52 137.158.152.199 TCP 39099 > webcache [FIN, ACK] Seq=0 Ack=283 Win=1728 Len=0 TSV=59340598 TSER=42207662 Frame 44 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39099 (39099), Dst Port: webcache (8080), Seq: 0, Ack: 283, Len: 0 No. Time Source Destination Protocol Info 45 17.647909 137.158.152.199 10.128.1.52 TCP webcache > 39099 [ACK] Seq=283 Ack=1 Win=33303 Len=0 TSV=42207662 TSER=59340598 Frame 45 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39099 (39099), Seq: 283, Ack: 1, Len: 0 No. Time Source Destination Protocol Info 46 17.677994 10.128.1.52 137.158.152.199 TCP 39100 > webcache [SYN] Seq=0 Len=0 MSS=1460 TSV=59340606 TSER=0 WS=2 Frame 46 (74 bytes on wire, 74 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 0, Len: 0 No. Time Source Destination Protocol Info 47 17.679170 137.158.152.199 10.128.1.52 TCP webcache > 39100 [SYN, ACK] Seq=0 Ack=1 Win=131070 Len=0 MSS=1460 WS=1 TSV=42207665 TSER=59340606 Frame 47 (78 bytes on wire, 78 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39100 (39100), Seq: 0, Ack: 1, Len: 0 No. Time Source Destination Protocol Info 48 17.679188 10.128.1.52 137.158.152.199 TCP 39100 > webcache [ACK] Seq=1 Ack=1 Win=5840 Len=0 TSV=59340606 TSER=42207665 Frame 48 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 1, Ack: 1, Len: 0 No. Time Source Destination Protocol Info 49 17.679241 10.128.1.52 137.158.152.199 TCP [TCP segment of a reassembled PDU] Frame 49 (703 bytes on wire, 703 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 1, Ack: 1, Len: 637 No. Time Source Destination Protocol Info 50 17.947952 10.128.1.52 137.158.152.199 TCP [TCP Retransmission] [TCP segment of a reassembled PDU] Frame 50 (703 bytes on wire, 703 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 1, Ack: 1, Len: 637 No. Time Source Destination Protocol Info 51 18.041756 137.158.152.199 10.128.1.52 TCP webcache > 39100 [ACK] Seq=1 Ack=638 Win=66608 Len=0 TSV=42207702 TSER=59340674 Frame 51 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39100 (39100), Seq: 1, Ack: 638, Len: 0 No. Time Source Destination Protocol Info 52 18.041778 10.128.1.52 137.158.152.199 HTTP POST http://wwwl.meebo.com/mcmd/events HTTP/1.1 (application/x-www-form-urlencoded) Frame 52 (154 bytes on wire, 154 bytes captured) Ethernet II, Src: Intel_7d:dc:49 (00:0c:f1:7d:dc:49), Dst: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb) Internet Protocol, Src: 10.128.1.52 (10.128.1.52), Dst: 137.158.152.199 (137.158.152.199) Transmission Control Protocol, Src Port: 39100 (39100), Dst Port: webcache (8080), Seq: 638, Ack: 1, Len: 88 [Reassembled TCP Segments (725 bytes): #50(637), #52(88)] Hypertext Transfer Protocol Line-based text data: application/x-www-form-urlencoded No. Time Source Destination Protocol Info 53 18.141936 137.158.152.199 10.128.1.52 TCP webcache > 39100 [ACK] Seq=1 Ack=726 Win=66608 Len=0 TSV=42207712 TSER=59340697 Frame 53 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Intel_7d:dc:49 (00:0c:f1:7d:dc:49) Internet Protocol, Src: 137.158.152.199 (137.158.152.199), Dst: 10.128.1.52 (10.128.1.52) Transmission Control Protocol, Src Port: webcache (8080), Dst Port: 39100 (39100), Seq: 1, Ack: 726, Len: 0 No. Time Source Destination Protocol Info 54 18.224398 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 54 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 55 18.549301 3com_4e:c5:cb Broadcast ARP Who has 10.128.1.34? Tell 10.128.255.254 Frame 55 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 3com_4e:c5:cb (00:0a:5e:4e:c5:cb), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 56 18.624740 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 56 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 57 18.784602 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 57 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 58 18.974370 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 58 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 59 19.491339 10.128.1.130 10.128.255.255 BROWSER Host Announcement COEOPNET, Workstation, Server, NT Workstation, Potential Browser Frame 59 (243 bytes on wire, 243 bytes captured) Ethernet II, Src: Micro-St_38:cf:d8 (00:13:d3:38:cf:d8), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.130 (10.128.1.130), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-dgm (138), Dst Port: netbios-dgm (138) NetBIOS Datagram Service SMB (Server Message Block Protocol) SMB MailSlot Protocol Microsoft Windows Browser Protocol No. Time Source Destination Protocol Info 60 19.724326 10.128.1.64 10.128.255.255 NBNS Name query NB WPAD.UCT.EXP.<00> Frame 60 (92 bytes on wire, 92 bytes captured) Ethernet II, Src: Foxconn_38:f6:2b (00:01:6c:38:f6:2b), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.64 (10.128.1.64), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-ns (137), Dst Port: netbios-ns (137) NetBIOS Name Service No. Time Source Destination Protocol Info 61 20.172354 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 61 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 62 20.322067 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 62 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 63 21.709846 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 63 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 64 21.869892 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 64 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 65 23.097494 10.128.1.41 10.128.255.255 BROWSER Host Announcement OLABISI, Workstation, Server, Print Queue Server, NT Workstation, Potential Browser Frame 65 (247 bytes on wire, 247 bytes captured) Ethernet II, Src: Intel_0e:98:95 (00:11:11:0e:98:95), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Internet Protocol, Src: 10.128.1.41 (10.128.1.41), Dst: 10.128.255.255 (10.128.255.255) User Datagram Protocol, Src Port: netbios-dgm (138), Dst Port: netbios-dgm (138) NetBIOS Datagram Service SMB (Server Message Block Protocol) SMB MailSlot Protocol Microsoft Windows Browser Protocol No. Time Source Destination Protocol Info 66 23.257370 Intel_4e:d8:94 Broadcast ARP Who has 10.128.0.114? Gratuitous ARP Frame 66 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:d8:94 (00:0e:0c:4e:d8:94), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP) No. Time Source Destination Protocol Info 67 23.417014 Intel_4e:c8:82 Broadcast ARP Who has 10.128.1.242? Gratuitous ARP Frame 67 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Intel_4e:c8:82 (00:0e:0c:4e:c8:82), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request/gratuitous ARP)

1 0

[Users] NAT ping best methods
by Jayesh Nambiar 27 Apr '07

27 Apr '07

Hi, I am using Openser 1.2 and all the UAs that register to it are only PAP2 or SPAs. But all these UAs are on a private network. I tried using the NAT ping method from Openser to keep the NAT port open, but still sometimes the NAT port gets blocked by some NAT boxes. I tried keeping the interval as 25 seconds. So then I tried sending Keep Alive Notify messages from the UAs itself towards my Openser, but there I see a strange problem. The problem is like the device does not send the notify message from the same port which it registered from in the first place. So the port which is stored in the locaion table gets blocked and again the user is not able to receive calls. The user is able to receive calls only after it registers again but then again it is inconsistent. Has anyone worked with PAP2 and Openser and has got some best method to maintain the NAT ports open so that the user receive calls consistently. I also wanted to know what is more efficient, the UAs sending keep-alives or Openser sending NAT-ping? Thanks in advance, w/regards, Jayesh Add online presence to your web pages, find out how here - http://messenger.yahoo.com/addpresence.php

2 1

[Users] openser: group module
by Helmut Kuper 27 Apr '07

27 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to call is_user_in funktion with a variable as group name. I want to create the group name on the fly, when a INVITE arrives. if(is_user_in("To", $var(a))) { ... } I got this error message: 0(27274) xl_parse_svname: parsing [$var(a)] 0(27274) xl_lookup_spec_name: found [fu] [18] 0(27274) find_cmd_export_t: found <xlog>(2) in module xlog [/home/hk/openser//lib/openser/modules/] 0(27274) parse error (479,27-34): syntax error 0(27274) parse error (479,34-35): bad arguments Is there a way to do dynamic group names für is_user_id? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGMNVj4tZeNddg3dwRApHgAJ4qP30SN9NgH5yzVzx3qDuawQskywCbBfE+ Uum6I3PsxVVrzFa/KvchnmA= =6hHJ -----END PGP SIGNATURE-----

2 3

[Serusers] Is there a bug in SERWeb ?
by inge 27 Apr '07

27 Apr '07

Hi all, I use SERWeb especially for call history. The core of this function is method.get_acc_entries.php. For some calls, I see that they are posted two times. During my investigation I see that the problem come from SQL request in method.get_acc_entries. In fact, it seems that when both calling and called party hang out in the same time, there are two BYE generated in acc table. Maybe the bug have been already reported, but I don't see nothing about this. Anyone one have an idea to correct it ? Thanks for your su

2 3

[Serusers] Where can I Download a soft phone?
by karl lu 27 Apr '07

27 Apr '07

Hi, I'm new here. I'm wondering if anyone knows where I can download a softphone, which will work with SER. Thank you very much, Karl

2 1

[Serusers] Ottendorf TM timers - global setting?
by Mahatma Sarasvati 27 Apr '07

27 Apr '07

Hello all, I see that one can set timeout variables per user in Ottendorf. Is it possible to avoid this and set them on a global or domain basis? My users aren't so particular and I'd like to avoid the effort of having to set the timers for every user. By the way, the default ser.cfg in Ottendorf is fantastic. Thanks for producing a well thought out config that demonstrates a good range of functionality. I've been using SER (+Asterisk) since 0.8.11 in my corporate environment (130 employees) so I'm not new, but I appreciate not having to take the time to sift though the code to figure out how the basics work. I'm eagerly awaiting more documentation on Ottendorf!! Thanks, Mahatma __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] WeSIP don't work with JDK1.6.0?
by GP.Coder 27 Apr '07

27 Apr '07

Hello,All I'm having problems running OpenSER(openser-1.2.0-notls_src.tar.gz) with WeSIP(wesip-02-04-07-beta-0.1.4.tar.gz) AS. SeasMessageEvent [SeasRouter] - New message received, from 192.168.0.119:3639 to 192.168.0.119:5060 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 57932,label=1021493454 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 192.168.0.119/3639 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:2001@192.168.0.200 SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:2001@192.168.0.200 SIP/2.0 SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:2001@192.168.0.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.119:15524 Max-Forwards: 69 From: "GP.Coder" <sip:5006@192.168.0.229>;tag=e495a30ed682466dad6c6fe809870efe;epid=1403e45675 To: <sip:2001@192.168.0.200> Call-ID: dec886aa3e954c6492b92fc8adef9de8(a)192.168.0.119 CSeq: 1 INVITE Contact: <sip:192.168.0.119:15524> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 479 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:57932 label:1021493454 into hashTable SipProcessor [SipProcessor[4]] - Transaction [null] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.0.229 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost SipProcessor [SipProcessor[4]] - error processing event... java.lang.NullPointerException at java.util.Hashtable.get(Hashtable.java:334) at com.voztele.sipservlet.sessions.StandardSipManager.findUncomplete(StandardSipManager.java:570) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeSipRequest(ConvergedHostValve.java:218) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeSIP(ConvergedHostValve.java:126) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:104) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:114) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.processor.SipProcessor.invokeContainer(SipProcessor.java:198) at com.voztele.sipservlet.processor.SipProcessor.processRequest(SipProcessor.java:142) at com.voztele.sipservlet.processor.SipProcessor.process(SipProcessor.java:272) at com.voztele.sipservlet.processor.SipProcessor.run(SipProcessor.java:253) at java.lang.Thread.run(Thread.java:619) SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] GP.Coder 2007-04-26

2 2

[Serusers] rtp proxy problem when re-inviting with changed local port in sdp
by Thomas Nilsen 27 Apr '07

27 Apr '07

Is it possible to get rtpproxy/nathelper to start a totally new rtpproxy session on re-invite and not use the previous pair of ports on the rtp proxy? In this case: A calls B behind nat. Sdp are rewritten to go though rtpproxy on two ports. X & Y. Some time goes by with conversation. Then the call chages codec, line or something (not important). In this case B ua changes rtp port locally also (incremented by 2). This is no problem. The re-invite is sent with new port. But problem now is that rtpproxy sees this and updates the sdp as always and it uses the previous X & Y port that was used on the rtpproxy before re-invite. The problem now is this. Packets from B to A goes correctly since A's local rtp port was same both before and after re-invite. But packets from A to B is arrivind on B's local port for the first sdp session, not the second. It seems that rtpproxy wont change this.. I assume this is a security measure for rtpproxy. But it should have some function to force new session ports on rtpproxy by parameters to force_rtpproxy() or something. As far as I can tell this is only a problem if the sip-ua is changing local port.. I thought maby calling unforce_rtp_proxy() before forcing rtp_proxy again would help, but don't work. Probably because of some internal working of nathelper. Anyone got an idea how to solve this? I would really appreciate any input on this.. - Thomas

1 0

RE: [Users] [OT] WeSIP don't work with JDK1.6.0?
by "C. Bergström" 26 Apr '07

26 Apr '07

I apologize for the noise/rant of this post, but I feel something needs to be said. "By now support will be made trough the USERS openser mailing list" Source: http://www.wesip.com/mediawiki/index.php/SupportMailingList 1) I don't object to you making a commercial product, but please do not direct your users to this list. 2) I'm no license expert and won't confirm this by downloading your software, but seems you are in violation of the GPL by linking into openser and not releasing your software under a compatible license. 3) "WeSIP is patent pending technology" Source: http://www.wesip.com/mediawiki/index.php/Main_Page Where exactly is this technology pending patent? Do you have a reference number by some chance? 4) I'm going to be a bit of a jerk and say you probably don't have the capital to successfully litigate any international licensing violations of your software. So why was this license chosen exactly? To point of this is to not waste more of everyone's time, but to encourage the company behind this seemingly fine product to GPL their source so others can contribute. Thanks C.

2 1

[Users] connect xmpp component to jabber xcp?
by Matt Winter 26 Apr '07

26 Apr '07

Hello, I'm having a problem connecting my openser xmpp module(in component mode) to a jabber xcp server. Has anyone done this? Here's my openser.cfg file. Thanks. # more /etc/openser/openser.cfg # # $Id: openser-xmpp.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script for XMPP GW # # make sure in your main SIP server that you send # only the adequate SIP MESSAGES to XMPP GW # # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 listen=udp:10.1.1.234:5060 alias=xmpp.jabbersrv.corp.com # ------------------ module loading ---------------------------------- mpath="/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # XMPP loadmodule "xmpp.so" modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "xmpp.jabbersrv.corp.com") modparam("xmpp", "xmpp_domain", "xmpp.jabbersrv.corp.com") modparam("xmpp", "xmpp_host", "jabbersrv.corp.com") modparam("xmpp", "backend", "component") modparam("xmpp", "xmpp_password", "BCD11234DC") # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo_xmpp") # -- usrloc params -- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; ### absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; }

2 4

[Users] sending NOTIFY messages
by Mark Price 26 Apr '07

26 Apr '07

Hi, Are there any generally used methods for sending SIP NOTIFY messages to a phone that is registered through openser? I.e. for sending events to sip phones? For example, I imagine a script that would register to the mydomain.org openser server as an administrative user and send a NOTIFY packet to user(a)mydomain.org. I'm hoping not to reinvent the wheel here. Mark Price

2 3

[Serusers] t_replicate "REGISTER" causes retransmission?
by Chahn Kim 26 Apr '07

26 Apr '07

Hi, I am testing simple(vs clean, per Jiri) t_replicate(server 2) on "REGISTER" from server 1 to another ser server 2. Server 2 logs "ERROR:registrar:update_contacts: invalid cseq for aor...". ngrep shows that server 1 send "REGISTER" twice. May be the re-transmitted request causes the incalid cseq? Can anyone tell me how to fix this problem? John Chahn Kim

1 0

[Users] Strange behaviour
by Ronnie Flink 26 Apr '07

26 Apr '07

Hi, We have openser (currently 1.1.1) deployed in our iptel-solution and almost everything seems to work ok but one strange thing has come up. A user has a few Dlink DVG-2001S (behind a net-hide firewall) that tries to register but doesn't succeed and my guess is that openser sends the reply to the incorrect port and there firewall blocks this. A small ngrep-dump of the session: U 2007/04/24 11:40:12.620330 xx.xx.243.11:3710 -> xx.xxx.248.10:5060 REGISTER sip:sip.domain.tld SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:5060;branch=z9hG4bK8294fce7242e1605. Max-Forwards: 70. From: xx55551904 <sip:xx55551904@sip.domain.tld>;tag=1c2bcb8b48c89388. To: xx55551904 <sip:xx55551904@sip.domain.tld>. Call-ID: fec625fb04d0f004(a)192.168.0.164. CSeq: 1 REGISTER. Contact: *. Expires: 0. Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS, INFO. Content-Length: 0. . U 2007/04/24 11:40:12.620555 xx.xxx.248.10:5060 -> xx.xxx.243.11:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.0.164:5060;branch=z9hG4bK8294fce7242e1605;received=xx.xxx.243.11. From: xx55551904 <sip:xx55551904@sip.domain.tld>;tag=1c2bcb8b48c89388. To: xx55551904 <sip:xx55551904@sip.domain.tld>. Call-ID: fec625fb04d0f004(a)192.168.0.164. CSeq: 1 REGISTER. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. Warning: 392 xx.xxx.248.10:5060 "Noisy feedback tells: pid=23636 req_src_ip=xx.xxx.243.11 req_src_port=3710 in_uri=sip:sip.domain.tld out_uri=sip :sip.domain.tld via_cnt==1". Does anyone have a guess what might be causing openser to send the reply to the wrong port ? The openser-box has a live internet-ip and and isn't behind any type of firewall or nat-device. I think all client nat-stuff is correct since it seems to work with other customers that are behind net-hide firewalls (broadband-routers). /Thanks in advance - Ronnie Flink

2 1

RE: [Serusers] global contact list
by Barta János 26 Apr '07

26 Apr '07

Hello! Thanks for your answer again, and I'm sorry for the delayed answer, but I was out of office. My biggest problem is, that I don't know, what I should see :-(. I have an eyeBeam window, that contains the "Calls & Conntats" parameters. In this window I can see my (user) contact list, but I didn't see the global one. For example (the PC is my user contact list): -PC -105 -12345 -barta and it is working fine, but I can not see here (in this window) the global one: sip:~# cat /var/www/xcap-root/rls-services/global/index <?xml version="1.0" encoding="UTF-8"?> <rls-services> <service uri="sip:bartaj-list@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> <service uri="sip:hr2@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> </rls-services> I think in that case I should see the next tree: -PC -105 -12345 -barta -Human resources -bartaj -12345 As You asked, I made a capture between the SER and XCAP. It was a little difficult, because I didn't see interesting traffic between the SER and XCAP, but a few minutes later I realised, that the SER for the communication (in that case) uses the loopback (lo) interface. You can see here the capture, that I made on the lo interface: 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [SYN] Seq=0 Len=0 MSS=16396 TSV=3808162073 TSER=0 WS=0 2 0.000056 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [SYN, ACK] Seq=0 Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162073 TSER=3808162073 WS=0 3 0.000065 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=1 Ack=1 Win=32767 Len=0 TSV=3808162073 TSER=3808162073 4 0.000072 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/pres-rules/users/12345/presence-rules.xml HTTP/1.1 5 0.000079 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [ACK] Seq=1 Ack=113 Win=32767 Len=0 TSV=3808162073 TSER=3808162073 6 0.000545 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK 7 0.000559 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 Ack=262 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 8 0.000571 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic 9 0.000578 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 Ack=870 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 10 0.005535 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [SYN] Seq=0 Len=0 MSS=16396 TSV=3808162079 TSER=0 WS=0 11 0.005556 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [SYN, ACK] Seq=0 Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162079 TSER=3808162079 WS=0 12 0.005565 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=1 Ack=1 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 13 0.005616 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/pres-rules/users/bartaj/presence-rules.xml HTTP/1.1 14 0.005623 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [ACK] Seq=1 Ack=114 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 15 0.005989 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK 16 0.006000 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 Ack=262 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 17 0.006011 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic 18 0.006017 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 Ack=870 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 19 2.979491 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/rls-services/global/index HTTP/1.1 20 2.979722 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK 21 2.979734 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 Ack=1126 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 22 2.979745 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic 23 2.979751 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 Ack=1887 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 You can see, that in the message number 19 the SER downloaded the index file. This file contains correct information: sip:~# wget 127.0.0.1/xcap/rls-services/global/index --17:22:43-- http://127.0.0.1/xcap/rls-services/global/index => `index' Connecting to 127.0.0.1:80... connected. HTTP request sent, awaiting response... 200 OK Length: 761 [text/plain] 100%[====================================>] 761 --.--K/s 17:22:43 (7.26 MB/s) - `index' saved [761/761] sip:~# sip:~# cat index <?xml version="1.0" encoding="UTF-8"?> <rls-services> <service uri="sip:bartaj-list@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> <service uri="sip:hr2@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> </rls-services> sip:~# And at last, how can I add query_rls_services and query_resource_list log informations into the config file? Thx, Jani -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Tuesday, April 03, 2007 11:18 AM To: Barta János Cc: serusers(a)iptel.org Subject: Re: [Serusers] global contact list Hi, this part of config file is not sufficient for me, but I guess that RLS related things stayed as in the original config you mentioned... In which way it doesn't work? (For example it returns non OK responses?) Please could you try to capture HTTP traffic between XCAP server and SER? (If SER really queries the right resource list document.) You can add more log information around important config parts (query_rls_services and query_resource_list functions are most important here). BTW the second message you sent is not subscription to resource list or user, it is subscription to event package 'sip-profile' (content of Event header field) which is not supported by SER. It is possible to use registration status as source for presence - in this case you have to have PA module on your server handling registrations and PA module parameter 'use_callbacks' must be set to 1. Vaclav <br><br>_____________________________________________________________________<br> Tudod már, hol nyaralsz? Úti beszámolókat, utazási ajánlatokat találsz a <a href="http://ad.adverticum.net/b/cl,1,6022,159388,223268/click.prm" target="new">travelline.hu-n!</a><br> Ha Te is megírod élményeidet, megnyerhetsz egy párizsi utat vagy egy kalandtúrát az Alpokba!

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[Serusers] REGISTERED(TRANSACTION) BUT NO DIALOG
by Adeyeye Michael 25 Apr '07

25 Apr '07

Hi, My clients get registered but no call is successfully transfered when I used UCTIMSCLIENT or Linphone. Besides I can't get any output or trace to know what is wrong with SER. micadeyeye@micadeyeye:/opt/OpenIMSCore$ sudo ngrep port 5060 -d eth0 Password: interface: eth0 (10.128.0.0/255.255.0.0) filter: (ip or ip6) and ( port 5060 ) PLS HELP. MIKE

1 0

[Serusers] No trace!!!
by Adeyeye Michael 25 Apr '07

25 Apr '07

Can anyone help out? I can see any output or trace. micadeyeye@micadeyeye:/opt/OpenIMSCore$ sudo ngrep port 5060 -d eth0 Password: interface: eth0 (10.128.0.0/255.255.0.0) filter: (ip or ip6) and ( port 5060 ) MIke.

1 0

[Users] WeSIP setup problem
by ventilo35＠free.fr 25 Apr '07

25 Apr '07

Hello I'm having problems running OpenSER with WeSIP AS. I have installed OpenSER 1.2 stable release, and also tried the latest version using SVN, and I use WeSIP 0.1.4. The same problem always arises: I get either "no route" if a servlet was invoked, or "481" if no servlet was run. In both cases, it seems WeSIP is not able to send the INVITE back to OpenSER. Here are the traces: <First trace: a servlet named neos is run; it just proxies the message> SeasMessageEvent [SeasRouter] - New message received, from 192.168.1.17:5060 to 192.168.1.17:5060 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 45959,label=642523441 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 192.168.1.17/5060 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:5002@192.168.1.11 SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:5002@192.168.1.11 SIP/2.0 SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:5002@192.168.1.11 SIP/2.0 Call-ID: 20011e80-296f38df-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005f00-296f38e8 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf80296f38e9 Contact: "5001" <sip:5001@192.168.1.17:5060> Max-Forwards: 69 User-Agent: LR SIP Phone Allow: INFO,UPDATE,NOTIFY,REFER Content-Type: application/sdp Content-Length: 406 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:45959 label:642523441 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKc0a814d51000bf80296f38e9] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.1.11 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHost [SipProcessor[4]] - Mapped request to:/neos ConvergedContextValve [SipProcessor[4]] - new session created :20011e80-296f38df-c0a814d5@192.168.1.11 doInvite within neos servlet: requestURI is sip:5002@192.168.1.11 StandardProxy [SipProcessor[4]] - sip:5002@192.168.1.11 Added to proxying targets. StandardProxy [SipProcessor[4]] - doProxy: proxying to sip:5002@192.168.1.11 StandardProxy [SipProcessor[4]] - ========== Request Sent ============== INVITE sip:5002@192.168.1.11 SIP/2.0 Call-ID: 20011e80-296f38df-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005f00-296f38e8 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.11:5060,SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf80296f38e9 Contact: "5001" <sip:5001@192.168.1.17:5060> Max-Forwards: 68 User-Agent: LR SIP Phone Allow: INFO,UPDATE,NOTIFY,REFER Content-Type: application/sdp Content-Length: 406 StandardProxy [SipProcessor[4]] - javax.sip.TransactionUnavailableException: no route! SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] <Second trace: the servlet is deactivated> SeasMessageEvent [SeasRouter] - New message received, from 192.168.1.17:5060 to 192.168.1.17:5060 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 32078,label=1509443196 flags=0 ExpressMessageChannel [ExpressMChannel[2]] - peerAddress = 192.168.1.17/5060 ExpressMessageChannel [ExpressMChannel[2]] - About to process INVITE sip:5002@192.168.1.11 SIP/2.0 SipConnector [ExpressMChannel[2]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[2]] - An incoming message is being assigned ExpressMessageChannel [ExpressMChannel[2]] - Done processing INVITE sip:5002@192.168.1.11 SIP/2.0 SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:5002@192.168.1.11 SIP/2.0 Call-ID: 20011b80-29725a1f-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005c00-29725a28 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf8029725a29 Contact: "5001" <sip:5001@192.168.1.17:5060> Max-Forwards: 69 User-Agent: LR SIP Phone Allow: INFO,UPDATE,NOTIFY,REFER Content-Type: application/sdp Content-Length: 408 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:32078 label:1509443196 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKc0a814d51000bf8029725a29] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.1.11 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHost [SipProcessor[4]] - Mapped request to:/neos ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Call-ID: 20011b80-29725a1f-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005c00-29725a28 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf8029725a29 Max-Forwards: 69 Content-Length: 0 I have thoroughly reviewed my configuration against the rules explained in the WeSIP website, everything seems in order (address OpenSER receives on is 192.168.1.11:5060, address for SEAS and WeSIP is 127.0.0.1:5080). Could you please tell me if this is something you already encountered and solved ? Thanks in advance Francois

2 1

[Users] XCAP Lite 0.9.1 released
by Adrian Georgescu 25 Apr '07

25 Apr '07

Change log xcap-lite (0.9.1) unstable; urgency=low * Added commit/rollback capability (use MDB2 database driver instead of DB) * Added inserted_time in watcher sync (diff provided by Klaus Darilion) The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.9.1.tar.gz Regards, Adrian

1 0

[Serusers] How to handle Cisco Phone registrations withouth DNS SRV records
by Mahatma Sarasvati 25 Apr '07

25 Apr '07

Hello All, I want to be able to handle registrations from Cisco Phones on the public internet to my SER server. I don't run my own public DNS servers. I use register.com for DNS and they don't support DNS SRV records as far as I know. So, my Cisco phone config has to have "voice.mydomain.com" in order for the REGISTER message to reach the SER server. However, I want my user accounts to be "jsmith(a)mydomain.com." The Cisco phone config is limiting because it appends the name of the proxy to the registration. Instead of 'username(a)mydomain.com' its trying to register with 'username(a)voice.mydomain.com' How can this be handled in the SER config? I use DNS SRV records on the LAN and it works great. What is the best method to handle this scenario? Thanks, Mahatma __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

RE: [Users] LCR in cache mode
by Papadopoulos Georgios 25 Apr '07

25 Apr '07

In my humble opinion this was a pretty useful relic from the past and it pretty much means I cannot upgrade to 1.2 :-( Are you suggesting that ip should be a primary key in the gw table? It doesn't make sense to me. Do you have any other suggestions to achieve what I am trying to do? thank you George > -----Original Message----- > From: Juha Heinanen [mailto:jh@tutpro.com] > Sent: Wednesday, April 25, 2007 12:54 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: RE: [Users] LCR in cache mode > > Papadopoulos Georgios writes: > > > Well, in my case the IP belongs to an asterisk server > which then sends > the call to 5 PSTN gateways. I use this > trick with the 5 different > prefixes to make asterisk > choose between the 5 PSTN gateways. It works > fine in db > mode but not in cache mode. Shouldn't lcr behave the same in > > both modes? > > db_mode does not exist anymore in 1.2. it was relic from the past. > > > I tried working around this by puting the 5 entries into 5 > different gw > groups (this sounds like a bigger hack to me > than what I am currently > doing). Still, in cache mode the > same entry is always picked. > > that is because according to the current design of lcr module > a gateway prefix is gateway specific gateways being > identified by their ip addresses. if request matches more > than one gateway, only one of them is left on the list and > the rest are pruned. > > -- juha > 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.

2 1

[Users] presence .... hard state?
by Cesc 25 Apr '07

25 Apr '07

Hi, RFC 3903 (PUBlishing to an event), talks about hard state and soft state. Soft state is properly handled by presence module ... that is, the one generated by PUB messages. But, is there any way to set hard state? the way I read it, it shall be possible to do: Bob subscribes to Alice presence info Alice has not published anything ... but we know we shall handle her state. We send to Bob a NOTIFY with the hard state (default) Or does the module already do that? how can I set the default hard state? Regards, Cesc

2 2

[Serusers] Using SER
by Adeyeye Michael 25 Apr '07

25 Apr '07

2 1

[Serusers] Bad Config file
by Adeyeye Michael 25 Apr '07

25 Apr '07

Hi, I am trying to use SER. At present, I added a line to ser.cfg in order to perform NAT transversal but I always get error on running my SER. ERROR: bad config file (1 errors) root@micadeyeye:/usr/src/sip_router# Pls find attached my ser.cfg file

2 1

RE: [Users] LCR in cache mode
by Papadopoulos Georgios 25 Apr '07

25 Apr '07

Hello Juha, Well, in my case the IP belongs to an asterisk server which then sends the call to 5 PSTN gateways. I use this trick with the 5 different prefixes to make asterisk choose between the 5 PSTN gateways. It works fine in db mode but not in cache mode. Shouldn't lcr behave the same in both modes? I tried working around this by puting the 5 entries into 5 different gw groups (this sounds like a bigger hack to me than what I am currently doing). Still, in cache mode the same entry is always picked. Regards, George > -----Original Message----- > From: Juha Heinanen [mailto:jh@tutpro.com] > Sent: Wednesday, April 25, 2007 11:43 AM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: [Users] LCR in cache mode > > Papadopoulos Georgios writes: > > > My gw table contains a gateway group with 5 entries which > have the same > IP but different prefixes. If I use lcr > module in cache mode > (db_mode=1), the same entry is always > picked. If I use lcr in db mode, > then it works as expected > and it picks randomly one of the five entries. > > Is this a known issue? My Openser version is 1.1.x > > the idea with gateway prefix is that prefix of a gateway is > added in front of the called number when the gateway is used. > thus it does not make sense to have the same gateway in the > same gw group many times with different prefixes. > > -- juha > 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.

2 1

[Users] Problem with xmpp module
by Antoine Fressancourt 25 Apr '07

25 Apr '07

Hello, I try to set up an IM gateway between a Jabber server (namely jabberd2) and OpenSER. I have not been a really original person as I have just followed the small tutorial given by Daniel-Constantin Mierla at FOSDEM 2007. But I have a small issue here. I have no problem sending messages from a jabber client to a SIP client but sometimes I can't send a message from my SIP client to the jabber client. I get the following trace : Apr 18 13:02:01 devlnx /usr/local/sbin/openser[12925]: message to XMPP : from <sip:8374@mydomain.test> to <sip:antoine* xmpp.mydomain.test(a)sip-xmpp.mydomain.test> Apr 18 13:02:01 devlnx /usr/local/sbin/openser[12925]: xmpp: invalid content-type 0x300ff Apr 18 13:02:02 devlnx /usr/local/sbin/openser[12930]: message to XMPP : from <sip:8374@mydomain.test> to <sip:antoine* xmpp.mydomain.test(a)sip-xmpp.mydomain.test> Apr 18 13:02:02 devlnx /usr/local/sbin/openser[12930]: xmpp: unable to write to command pipe: Bad file descriptor >From time to time I get an error message saying : Apr 18 12:04:01 devlnx /usr/local/sbin/openser[12440]: WARNING: script writer didn't release transaction When i send a message successfully I get : Apr 18 12:47:32 devlnx /usr/local/sbin/openser[12718]: message to XMPP : from <sip:8374@mydomain.test> to <sip:antoine* xmpp.mydomain.test(a)sip-xmpp.mydomain.test> Apr 18 12:47:32 devlnx /usr/local/sbin/openser[12718]: xmpp: invalid content-type 0x300ff Apr 18 12:47:33 devlnx /usr/local/sbin/openser[12715]: message to XMPP : from <sip:8374@mydomain.test> to <sip:antoine* xmpp.mydomain.test(a)sip-xmpp.mydomain.test> My configuration file is the classic, out of the box openser.cfg for a proxy with the configuration described in the FOSDEM presentation included before the uri checking. Do you have an idea of what the problem can be, and how I can solve this ? Thank you in advance for your answers, Antoine

3 3

[Users] LCR in cache mode
by Papadopoulos Georgios 25 Apr '07

25 Apr '07

Hello all, My gw table contains a gateway group with 5 entries which have the same IP but different prefixes. If I use lcr module in cache mode (db_mode=1), the same entry is always picked. If I use lcr in db mode, then it works as expected and it picks randomly one of the five entries. Is this a known issue? My Openser version is 1.1.x thank you 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.

2 1

[Users] radius - multi leg accounting
by Rosario Pingaro 25 Apr '07

25 Apr '07

Hi list I need to implement multileg radius accounting on openser 1.1.x cfg. I have configured radius to handle Sip-Leg-Source and Sip-Leg-Destination. The problem is to configure openser.cfg propely. In the beginning i have: modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) then in the route that handle the invite I load the avp by radius: avp_load_radius("callee"); But then I miss the point to have two accounting start and two accounting stop, one for each leg. Then...if it is not possible to have two accounting records, I would liek to know how to store all the information of the two legs insede one start and one stop. something like this one? avp_write("$from/username", "$avp(i:110)"); avp_pushto("$ru","$avp(s:callee_callfwd)"); avp_write("$ruri/username", "$avp(i:111)"); Hope some one can drive me to understand better this situation. Regards Rosario

2 1

[Serusers] New way to manage your configuration files: SER configuration build system
by Greger V. Teigre 25 Apr '07

25 Apr '07

SER's configuration files can be quite complex, both to understand as a newcomer and to manage for experts. The challenge is to make getting started easy, while keeping the power of the configuration language. Thus, as an example, some would like to simplify ser.cfg so they can use: if(nated) { fix_nating() } while others need a lot more control. The SER - Getting Started configuration files and document quickly became a starting point for newcomers. Also, as you normally modify and add to your first configuration, the Getting Started files have formed the basis for many configuration files in production. We realized that updating and adding functionality to the Getting Started configurations quickly became time-consuming. So, we needed a better way of managing and improving these reference configurations. The result of these needs is a new build system for SER configuration files where you use a configure script to generate your basic configurations and run make to create the ser.cfg configuration file. Once you have run configure, you can edit some (simple) configuration files to change how ser.cfg is created. You can also add code snippets of your own. In order to make this system something people can grow with (and something experts would like to migrate their configs to), the build system is built around a generic splitting of ser.cfg into smaller pieces, so that different logical parts of your SER config can be managed in separate files. In its simplest form, the build system is just an empty structure of files you can use to manage your ser.cfg code. In its most advanced form, you can use code from the Getting Started files and add your own code and even modify the way things work (this last part is not yet ready in the initial version of the build system). The idea is that migration of your ser.cfg will become easier if you use the build system. Finally, a logging system has been included. It offers a powerful way to unify your logging across your ser.cfg, as well as multiple ways of controlling when logging is done (you can for example turn off logging for a production server, both by removing logging code and by dynamically turning it on and off). Well, I think that's enough for an introduction. The announcement can be found here: http://www.iptel.org/finally_making_ser_accessible_to_newcomers_while_keepi… while the build system documentation (and actual code until it gets into CVS) can be found at: http://iptel.org/ser/doc/buildsystem Play with it, test it, and please send your comments, ideas, suggestions to serdev or serusers! g-)

1 0

RE: [Users] Malformed Radius packet
by Edson 24 Apr '07

24 Apr '07

Sorry... The contents follows: Radius Protocol Code: Accounting-Request (4) Packet identifier: 0x61 (97) Length: 128 Authenticator: 22B8D8F7AEE79429749AED94FBC1874C Attribute Value Pairs AVP: l=6 t=Acct-Status-Type(40): Start(1) AVP: l=6 t=Service-Type(6): IAPP-Register(15) AVP: l=6 t=Unknown-Attribute(102): 000000C8 AVP: l=6 t=Error-Cause(101): Unknown(1) [Not enough room in packet for AVP] Looking on the dictionary, there is no attribute 102... neither error-cause 101 or attribute value of C8. Any ideas where they are? Edson. > -----Original Message----- > From: Edson [mailto:4lists@gmail.com] > Sent: terça-feira, 24 de abril de 2007 15:10 > To: 'Juha Heinanen' > Cc: 'users(a)openser.org' > Subject: RE: [Users] Malformed Radius packet > > Humm.... > > This is what appears: > > > Edson. > ------ > "It wouldn't be sell if it were easy..." (myself) > > > > -----Original Message----- > > From: Juha Heinanen [mailto:jh@tutpro.com] > > Sent: terça-feira, 24 de abril de 2007 14:51 > > To: Edson > > Cc: users(a)openser.org > > Subject: [Users] Malformed Radius packet > > > > Edson writes: > > > > > Error: WARNING: Malformed RADIUS packet from host 127.0.0.1: packet > > > attributes do NOT exactly fill the packet > > > > can you repeat this error at will? if so, check using wireshark how the > > packet looks like. > > > > -- juha

3 2

[Users] Malformed Radius packet
by Edson 24 Apr '07

24 Apr '07

Hi... I'm receiving this message on FreeRadius (1.1.3), running RadiusClient-NG (0.5.5.1) and OpenSER SVN 20070416-143000 on an OpenSUSE 10.2-64 bits: Error: WARNING: Malformed RADIUS packet from host 127.0.0.1: packet attributes do NOT exactly fill the packet Since the dictionary used by the RadiusClient-NG and FreeRadius are the same, any ideas from the source of this error? The dictionary is the one provided on the Radius module, and I INCLUDE it on FreeRadius AND RadiusClient-NG. Edson.

3 3

[Serusers] Latest SER CVS sources compilation avpops module error
by tzieleniewski 24 Apr '07

24 Apr '07

Hi, I downloaded the latest CVS trunk sources and got the following error: make[1]: Entering the directory `/home/ser/src/ser-trunk/modules/avpops' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev2"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -c avpops.c -o avpops.o In file included from avpops_impl.h:41, from avpops_parse.h:38, from avpops.c:50: avpops_db.h:66: error: expected ‘)’ before ‘*’ token make[1]: *** [avpops.o] Error 1 make[1]: Leaving the directory `/home/ser/src/ser-trunk/modules/avpops' make: *** [modules] Error 1 Cheers Tomasz

4 3

[Users] mysql "dual master" problem
by Chahn Kim 24 Apr '07

24 Apr '07

Hi, all. I am trying to setup two openser servers each running its own Mysql with dual master configuration. Master 1 has initially all locations info in its table. Once master 2 comes on as mysql dual master, It seems functioning initially however, as time goes by, I noticed that location table changes in a way that master 1 loses all location entries when those devices register to master 2. Eventually only locations where master 2 responded to "REGISTER" are being in the database. Can anyone tell me what may be going wrong? John Kim

3 2

[Users] Need Help mi_fifo: t_uac_dlg
by bogos dan-cristian 24 Apr '07

24 Apr '07

Guys, since I have upgraded to 1.2 my fifo interface stopped working. # MI_Fifo Module Parameters modparam("mi_fifo","fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0666) Example of old command I was using in 1.1: cat 1177428091.96-1.14-4263 (using # to mark the command, so not used in the command) ########################################## :t_uac_dlg:dan_fifo NOTIFY sip:jksdjksjdksjdkj@domain.com . From: sip:jksdjksjdksjdkj@domain.com;tag=3152259304 To: <sip:jksdjksjdksjdkj@domain.com>;tag=babble Event: accounting Content-Type: text/xml; charset=UTF-8 Call-ID: 3827705921(a)10.0.0.4 CSeq: 20 NOTIFY . <?xml version="1.0"?> <accountingInfo> <credit>0.00</credit> </accountingInfo> . #################################### Anybody can help me with an example of using t_uac_dlg with mi_fifo? I get differet errors: command empty, pharse error before <?xml line .. etc, and I am forced to downgrade otherwise. Thank you in advance, Dan Bogos __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] need help!
by Mukesh Prajapati 24 Apr '07

24 Apr '07

Hi all, frm where i can get latest rpm package for ser. in ftp of ser there is only debian packages. waiting for reply, thnx mukesh --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

1 0

[Users] Unconditional Call Forward help...
by Geir O. Jensen 24 Apr '07

24 Apr '07

I need CFU on our systems, however - it seems to work but if the forwarded call isn't picked up within a time limit the system just drops the call. My script is something like this; (...) loadmodule "tm.so" modparam ("tm", "fr_timer", 30) modparam ("tm", "fr_inv_timer", 10) route{ xlog("L_INFO", "Route - $fu $rm $ru"); t_on_failure("1"); (... Some tweaking ...) t_relay(); } failure_route[1] { if (!t_was_cancelled()) { xlog("L_INFO","Failure_route[1] - redirect to callcentre"); seturi("sip:callcentre@localhost"); append_branch(); t_relay(); }; return; } So what happens is this; I call a number... After 10 seconds the call "fails" and failure_route is invoked, whereupon the call is "redirected" to the new uri. It rings for 10 seconds and then the the call is just terminated... (no call to failure_route)... The log looks like this; 15:30:00 Route sip:usera@localhost INVITE sip:userb@localhost 15:30:10 Failure_route[1] - redirect to callcentre 15:30:20 Route sip:usera@localhost ACK sip:userb@localhost Any ideas? Typically I would like for the 2dn call to ring indefinitely... // sip:gojensen@uninett.no | h323: 004710012 pstn: +47 73 55 79 23 | fax: +47 73 55 79 01

3 3

[Serusers] Inbound routes to SIP
by Edsel Pagunsan 24 Apr '07

24 Apr '07

Hi, Can anybody send a sample config of ser.cfg for inbound routes which will be coming from asterisk server? UA1------->Asterisk----------->SER-------->UA2. Outbound is working perfectly. Thank you very much. Bengie888 --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

1 0

[Users] rtpproxy - ICMP unreacheable host- FINALLY SOLVED
by Amanda Mattiuz 24 Apr '07

24 Apr '07

Hi! I'd like to thank all for the help!! I solved my problem... in fact the problem was that my server had this firewall running in another machine and I was not aware of that, so I was checking my IP tables and everything seem to be opened, but in fact it was not. That's the reason for the "unreacheable destination". The unreacheable destination was, in fact, the server machine itself where I have also rtpproxy running.... the cfg file and everything is OK. Thansk again for the help! Amanda On 4/16/07, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > Hi Mattiuz , > > > Please follow the step bu steps in SDP section when invites and 200 Ok > Responsing from UAC and UAS > > ie.. > in fixed_nated_sdp( flag) test one by one from 1 to 12 flag, > > Because of Some Fire will blocks the SDP messages of mail in Contact > -header ... and other o and c and also m option in Sdp message body. > > > > > On 4/14/07, Amanda Mattiuz <amattiuz(a)gmail.com> wrote: > > > > Just another question (for today at least.... ;-) ) : > > > > Does anybody have a client that works with RTPproxy and Openser that I > > can use?? Maybe the problem is in my client, I don't know.... so It would be > > nice if I could try with anotehr client that I know that works... > > > > Thanks again, > > Amanda > > > > On 4/3/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > wrote: > > > > > > Hi Amanda, > > > > > > be sure the devices are symmetric from RTP point of view - otherwise > > > the > > > traversal will not work. Also note that first you need incoming RTP > > > (from client to RTPproxy) in order to be able to send RTP behind the > > > nat. > > > > > > regards, > > > bogdan > > > > > > Amanda Mattiuz wrote: > > > > Hello! > > > > > > > > > > > > I'm using rtpproxy to handle NAT, but right now I have problems of > > > > unreacheable host... I believe client's firewalls are blocking UDP > > > > traffic coming from the proxy ... I tried to change port range and > > > > nothing seem to make it work... > > > > > > > > I was wondering if any of you kows something that may fix it.... > > > > > > > > Thanks > > > > > > > > -- > > > > ------------------------------------------------------------------- > > > > Amanda Remes Mattiuz > > > > amattiuz(a)gmail.com <mailto: amattiuz(a)gmail.com> > > > > > > > ------------------------------------------------------------------------ > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users(a)openser.org > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > -- > > ------------------------------------------------------------------- > > Amanda Remes Mattiuz > > amattiuz(a)gmail.com > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > www.hyperion-tech.com > Client and Parent company :- www.august-networks.com > -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

1 0

[Users] typo error in acc_mod.c
by Helmut Kuper 24 Apr '07

24 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, there is a typo error in acc_mod.c replace line 222 ({"acc_totag_column", STR_PARAM, &acc_totag_col },) with ({"acc_to_tag_column", STR_PARAM, &acc_totag_col },) this will make modparam acc_to_tag_column work. Alternatively u can use acc_totag_column in openser.cfg as modparam name. regard Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGLeMz4tZeNddg3dwRAhhHAJ9eeXdpVfWc/SxPnMl7AZPV1NzvGACdEKIv JTj1E8XYptFYPB0HG4K92Jw= =DudI -----END PGP SIGNATURE-----

2 1

[Users] FIFO chmod
by Joao Pereira 24 Apr '07

24 Apr '07

Hello My serweb needs the FIFO to be with writing permissions: prwxrwxrwx 1 root root 0 Apr 23 23:23 /tmp/openser_fifo But every time I reload Openser, the openser_fifo permissions dont allow serweb to use the FIFO. In the old SER I did: fifo_mode=0666 But this doesn't work in OpenSER Is there any similar command in Openser? Thanks Regards Joao Pereira

2 1

[Users] ACC module, always prints message for successful transaction
by Tim Madorma 24 Apr '07

24 Apr '07

Hi, I would rather not have a syslog message for successful transactions - just failed ones. I have set the appropriate flags for the transactions that I would like to see: my settings are: modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) But in my syslog, I get a log entry for every successful transaction - such as: Apr 23 13:54:12 homer openser[17280]: [ID 729047 local0.notice] ACC: transaction answered: timestamp=1177350852;method=INVITE;from_t ag=188229;to_tag=6110;call_id=188229-14338(a)65.185.233.183;code=200;reason=OK How do I stop ACC from logging for successful transactions? thanks, Tim

2 1

[Serusers] SER + RTPproxy with video
by roayu 24 Apr '07

24 Apr '07

Good day! I'm a newbie on SER. I had configured a SER server running with RTPproxy. I'm using SER v0.9.6 and RTPproxy Basic Version 20040107. It works fine with the audio but not for the video. From what i understand that this version of rtpproxy do support both audio and video. I had trace the message header of SDP and it seems fine. I'm not sure what caused this happended. Could you pls guide me on this? Thanks. Regards, Roa

1 0

Re: [Users] memory issues
by Daniel-Constantin Mierla 23 Apr '07

23 Apr '07

Hello, daily snapshots are enabled now for 1.2.x as well: http://www.openser.org/downloads/snapshots/openser-1.2.x/ Cheers, Daniel On 04/23/07 17:47, Ovidiu Sas wrote: > Hi Tim, > > Check the download page from openser website: > http://www.openser.org/mos/view/Download/: > > The command that you need to run: > svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 > openser > > Make sure that you have svn installed. > > > Regards, > Ovidiu Sas > > On 4/23/07, Tim Madorma <tmadorma(a)gmail.com> wrote: >> Hi Daniel, >> >> I have run into a leak in 1.2 and I assume it is the same one that >> Ovidiu ran into. I see in your response that it was "backported to >> 1.2", but I'm not sure how to get the fix. When I look at the SVN >> repository at: >> http://www.openser.org/pub/openser/latest-1.2.x/, the date is earlier >> than the date of your email exchange so I don't think the fix has been >> added there. Can you please let me know how I can get it? >> >> thanks, >> Tim >> >> On 3/23/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> > Hello Ovidiu, >> > >> > On 03/23/07 17:04, Ovidiu Sas wrote: >> > > Hi Daniel, >> > > >> > > Can we backport this one to 1.2? >> > already done, two minutes after the commit in trunk. >> > >> > Cheers, >> > Daniel >> > >> > > >> > > >> > > Regards, >> > > Ovidiu Sas >> > > >> > > On 3/22/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> > >> Hello, >> > >> >> > >> the supposed fragmentation turned out to be a mem leak in pkg. >> Please >> > >> take the latest SVN version and try again to see if you got same >> > >> results. >> > >> >> > >> Thanks, >> > >> Daniel >> > >> >> > >> On 03/19/07 18:52, Christian Schlatter wrote: >> > >> > ... >> > >> >>> The memory statistics indeed show a high number of memory >> fragments: >> > >> >>> >> > >> >>> before 'out of memory': >> > >> >>> >> > >> >>> shmem:total_size = 536870912 >> > >> >>> shmem:used_size = 59607040 >> > >> >>> shmem:real_used_size = 60106488 >> > >> >>> shmem:max_used_size = 68261536 >> > >> >>> shmem:free_size = 476764424 >> > >> >>> shmem:fragments = 9897 >> > >> >>> >> > >> >>> after 'out of memory' (about 8000 calls per process): >> > >> >>> >> > >> >>> shmem:total_size = 536870912 >> > >> >>> shmem:used_size = 4171160 >> > >> >>> shmem:real_used_size = 4670744 >> > >> >>> shmem:max_used_size = 68261536 >> > >> >>> shmem:free_size = 532200168 >> > >> >>> shmem:fragments = 57902 >> > >> >>> >> > >> >>>> >> > >> >>>> You can try to compile openser with -DQM_JOIN_FREE (add it >> in DEFS >> > >> >>>> variable of Makefile.defs) and test again. Free fragments >> should be >> > >> >>>> merged and fragmentation should not occur -- processing >> will be >> > >> >>>> slower. We will try for next release to provide a better >> solution >> > >> >>>> for that. >> > >> >>> >> > >> >>> Compiling openser with -DQM_JOIN_FREE did not help. I'm not >> sure how >> > >> >>> big of a problem this fragmentation issue is. >> > >> >> What is the number of fragments with QM_JOIN_FREE after >> flooding? >> > >> > >> > >> > The numbers included above are with QM_JOIN_FREE enabled. >> > >> > >> > >> >>> Do you think it would make sense to restart our production >> openser >> > >> >>> instances from time to time just to make sure they're not >> running >> > >> >>> into this memory fragmentation limits? >> > >> >> The issue will occur only when the call rate reaches the >> limits of >> > >> >> the proxy's memory. Otherwise the chunks are reused. >> Transactions and >> > >> >> avps are rounded up to be sure there will be minimized the >> number of >> > >> >> different sizes for memory chunks. It wasn't reported too often, >> > >> >> maybe that's why no big attention was paid to it. This memory >> system >> > >> >> is in place since the beginning of ser. Alternative is to use >> sysv >> > >> >> shared memory, but is much slower, along with libc private >> memory >> > >> >> manager. >> > >> > >> > >> > I've done some more testing and the same out-of-memory stuff >> happens >> > >> > when I run sipp with 10 calls per second only. I tested with >> > >> > 'children=1' and I only could get through about 8200 calls (again >> > >> > those 8000 calls / process). And this is with QM_JOIN_FREE >> enabled. >> > >> > >> > >> > Memory statistics: >> > >> > >> > >> > before: >> > >> > shmem:total_size = 536870912 >> > >> > shmem:used_size = 2311976 >> > >> > shmem:real_used_size = 2335720 >> > >> > shmem:max_used_size = 2465816 >> > >> > shmem:free_size = 534535192 >> > >> > shmem:fragments = 183 >> > >> > >> > >> > after: >> > >> > shmem:total_size = 536870912 >> > >> > shmem:used_size = 1853472 >> > >> > shmem:real_used_size = 1877224 >> > >> > shmem:max_used_size = 2465816 >> > >> > shmem:free_size = 534993688 >> > >> > shmem:fragments = 547 >> > >> > >> > >> > So I'm not sure if this is really a fragmentation issue. 10 >> cps surely >> > >> > doesn't reach the proxy's memory. >> > >> > >> > >> > Thoughts? >> > >> > >> > >> > Christian >> > >> > >> > >> > >> > >> > >> > >> >> Cheers, >> > >> >> Daniel >> > >> >> >> > >> >>> >> > >> >>> thanks, >> > >> >>> Christian >> > >> >>> >> > >> >>>> >> > >> >>>> Cheers, >> > >> >>>> Daniel >> > >> >>>> >> > >> >>>> On 03/18/07 01:21, Christian Schlatter wrote: >> > >> >>>>> Christian Schlatter wrote: >> > >> >>>>> ... >> > >> >>>>>> >> > >> >>>>>> I always had 768MB shared memory configured though, so I >> still >> > >> >>>>>> can't explain the memory allocation errors I got. Some >> more test >> > >> >>>>>> runs revealed that I only get these errors when using a more >> > >> >>>>>> production oriented config that loads more modules than >> the one >> > >> >>>>>> posted in my earlier email. I now try to figure out what >> exactly >> > >> >>>>>> causes these memory allocation errors that happen >> reproducibly >> > >> >>>>>> after about 220s at 400 cps. >> > >> >>>>> >> > >> >>>>> I think I found the cause for the memory allocation >> errors. As >> > >> >>>>> soon as I include an AVP write operation in the routing >> script, I >> > >> >>>>> get 'out of memory' messages after a certain number of calls >> > >> >>>>> generated with sipp. >> > >> >>>>> >> > >> >>>>> The routing script to reproduce this behavior looks like >> (full >> > >> >>>>> config available at >> > >> >>>>> http://www.unc.edu/~cschlatt/openser/openser.cfg): >> > >> >>>>> >> > >> >>>>> route{ >> > >> >>>>> $avp(s:ct) = $ct; # commenting this line solves >> > >> >>>>> # the memory problem >> > >> >>>>> >> > >> >>>>> if (!method=="REGISTER") record_route(); >> > >> >>>>> if (loose_route()) route(1); >> > >> >>>>> >> > >> >>>>> if (uri==myself) rewritehost("xx.xx.xx.xx"); >> > >> >>>>> route(1); >> > >> >>>>> } >> > >> >>>>> >> > >> >>>>> route[1] { >> > >> >>>>> if (!t_relay()) sl_reply_error(); >> > >> >>>>> exit; >> > >> >>>>> } >> > >> >>>>> >> > >> >>>>> An example log file showing the 'out of memory' messages is >> > >> >>>>> available at >> http://www.unc.edu/~cschlatt/openser/openser.log . >> > >> >>>>> >> > >> >>>>> Some observations: >> > >> >>>>> >> > >> >>>>> - The 'out of memory' messages always appear after about >> 8000 test >> > >> >>>>> calls per worker process. One call consists of two SIP >> > >> >>>>> transactions and six end-to-end SIP messages. An openser >> with 8 >> > >> >>>>> children handles about 64'000 calls, whereas 4 children only >> > >> >>>>> handle about 32'000 calls. The sipp call rate doesn't >> matter, only >> > >> >>>>> number of calls. >> > >> >>>>> >> > >> >>>>> - The 8000 calls per worker process are independent from the >> > >> >>>>> amount of shared memory available. Running openser with -m >> 128 or >> > >> >>>>> -m 768 does not make a difference. >> > >> >>>>> >> > >> >>>>> - The more AVP writes are done in the script, the less >> calls go >> > >> >>>>> through. It looks like each AVP write is leaking memory >> (unnoticed >> > >> >>>>> by the memory statistics). >> > >> >>>>> >> > >> >>>>> - The fifo memory statistics do not reflect the 'out of >> memory' >> > >> >>>>> syslog messages. Even if openser does not route a single SIP >> > >> >>>>> message because of memory issues, the statistics still >> show a lot >> > >> >>>>> of 'free' memory. >> > >> >>>>> >> > >> >>>>> >> > >> >>>>> All tests were done with openser SVN 1.2 branch on Ubuntu >> dapper >> > >> >>>>> x86. I think the same is true for 1.1 version but I >> haven't tested >> > >> >>>>> that yet. >> > >> >>>>> >> > >> >>>>> >> > >> >>>>> Christian >> > >> >>>>> >> > >> >>> >> > >> >>> >> > >> > >> > >> > >> > >> > _______________________________________________ >> > >> > Users mailing list >> > >> > Users(a)openser.org >> > >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> > >> > >> >> > >> _______________________________________________ >> > >> Users mailing list >> > >> Users(a)openser.org >> > >> http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> > > >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

3 4

[Serusers] Mediaproxy sip-asymmetric-clients
by Ali Zaidi 23 Apr '07

23 Apr '07

Hi, I have been using ser with mediaproxy for more than a year without any major issue. I have couple of asymmetric UAs, and i'm not able to call them most of the time. i noticed in my ser.cfg following lines. My ser.cfg =========================== modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") # modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-asymmetric-clients") # modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-asymmetric-clients") =========================== I have above files in ser folder, and i tried to put user agent in sip-clients file, i don't know if i'm doing it right. Ii'm still trying to figure it out, what should i put in these sip-clients and rtp-clients files (username? URI ), and how do i know if they are working. I just put UA 5007860215 and restarted ser and mediaproxy but ngrep are same. my sip-clients file is as follows: ########### START ################## # 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 SIP signaling. For clients that are asymmetric regarding the RTP # media streams, use the equivalent RTP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # ipDialog SipTone.* #<== what is it? 5007860215 ########### END OF FILE############ client sending keep alive message, reason i setup keep alive on my client is becouse they were keep loosing registration, so that problem is solved. Following asymentric client coming through port 50005 and SER sending reply at port 5060. ################# U XX.XX.55.170:50005 -> XX.XX.66.231:5060 NOTIFY sip:ser.mydomain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK-2173a014..From: Tony < sip:5007860215@ser.mydomain.com>;tag=a11c95e2a205caa8o0..To: <sip: ser.mydomain.com>..Call-ID: 9b70d42-e62ebac8(a)192.168.1. 25..CSeq: 4043 NOTIFY..Max-Forwards: 70..Event: keep-alive..User-Agent: Sipura/SPA1001- 3.1.8(SEc)..Content-Length: 0. ... # U XX.XX.66.231:5060 -> XX.XX.55.170:5060 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 192.168.1.25:5060 ;branch=z9hG4bK-ceived=XX.XX.55.170..From: Tony <sip:5007860215@ser.mydomain.com>;tag=a11c95e2a205caa8o0..To: <sip:ser.mydomain.com>;tag=e4b88024d5465a32af4d4d872ac6bb61.0ce2..Call-ID: 9b70d42-e62ebac8@192.168.1.25..CSeq: 4043 NOTIFY..Server: Sip EXpress router (0.9.5-pre4 (i386/linux))..Content-Length: 0..Warning: 392 ser.mydomain.com:5060"Noisy feedback tells: pid=14718 req_src_ip= 70.18.55.170 req_src_port=50005 in_uri=sip:ser.mydomain.com out_uri=sip: ser.mydomain.com via_cnt==1".... ################ If i ping my UA with (serctl ping sip:5007860208@ser.mydomain.com) i get following ngrep. It appears that SER has no problem reaching UA at port 50005. ################### U XX.XX.66.231:5060 -> XX.XX.55.170:50005 OPTIONS sip:5007860215@192.168.1.25:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:XX.XX.66.231;ftag=887df13cdb9661be5c6769ac08d69940-385f;lr=on>..Via: SIP/2.0/UDP XX.XX.66.231;branch=z9hG4bK88b5.16ac1512.0..Via: SIP/2.0/UDP 67.83.66.231;branch=z9hG4bK88b5.06ac1512.0..To: <sip:5007860215@ser.igranet.com >..From: sip:daemon@ool-435342e7.covad.net;tag=887df13cdb9661be5c6769ac08d69940-385f..CSeq: 10 OPTIONS..Call-ID: 7effbd40-0@XX.XX.66.231..Content-Length: 0..User-Agent: Sip EXpress router(0.9.5-pre4 (i386/linux))..Contact:< sip:daemon@XX.XX.66.231:5060>.... # U XX.XX.55.170:50005 -> XX.XX.66.231:5060 SIP/2.0 200 OK..To: < sip:5007860215@ser.igranet.com>;tag=8b659eab4f39640i0..From: sip:daemon@ool-435342e7.covad.net;tag=887df13cdb9661be5c6769ac08d69940-385f..Call-ID: 7effbd40-0@XX.XX.66.231..CSeq: 10 OPTIONS..Via: SIP/2.0/UD P XX.XX.66.231;branch=z9hG4bK88b5.16ac1512.0..Via: SIP/2.0/UDP XX.XX.66.231;branch=z9hG4bK88b5.06ac1512.0..Record-Rou te: <sip:XX.XX.66.231;ftag=887df13cdb9661be5c6769ac08d69940-385f;lr=on>..Server: Sipura/SPA1001-3.1.8(SEc)..Content-L ength: 0..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura.... ################## please help... Ali....

2 3

[Users] MediaProxy Scenario
by Edson 23 Apr '07

23 Apr '07

Hi... I have two questions (well, three): 1a) How should Mediaproxy be configured so to permit a call interruption? 1b) Is there any external interface that permit communications with the Mediaproxy process? 2) How can a call (RTP traffic) be recorded with MediaProxy? Any format... Edson. ------ "It wouldn't be sell if it were easy..."

1 0

[Users] Openser with unixODBC + sqlite driver
by Sylvain Boily 23 Apr '07

23 Apr '07

Hi, I would like to connect openser 1.2 to sqlite db with unixodbc. I have installed unixodbc et sqlite driver and i configure openser.cfg but it doesn't working. I seems the configuration wants user:password@ip but for sqlite there no need this. I have testing lot off configuration but it doesn't work. My configuration and log : File /etc/odbc.ini [mysqlitedb] Description = SQLite Driver = SQLite Database = /tmp/mytest.db Timeout = 100000 StepAPI = No NoWCHAR = No LongNames = No File /etc/odbcinst.ini [SQLite] Description = SQLite ODBC Driver Driver = /usr/lib/odbc/libsqliteodbc.so Setup = /usr/lib/odbc/libsqliteodbc.so UsageCount = 1 File /etc/openser/openser.cfg loadmodule "unixodbc.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc|auth_db", "db_url","unixodbc://SQLite/mysqlitedb") My log : Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: AUTH module - initializing Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: AUTH_DB module - initializing Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: ERROR:unixodbc:new_connection: failed to connect Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: register_udomain(): Can not open database connection Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: domain_fixup(): Error while registering domain Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 148 Apr 21 13:33:25 sylvain-desktop /usr/sbin/openser[24539]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Thank for your help. Sylvain

3 4

[Serusers] IMPORTANT: Invitation to discuss the management of iptel.org
by Greger V. Teigre 23 Apr '07

23 Apr '07

Dear all, The way decisions has been made for SER and iptel.org is not explicitly described anywhere. In fact, there has not really been an official policy for how to make decisions. The consequence has been that some decisions have come out of consensus on the mailing lists (normally the developer mailing lists), some have come out of developers/individuals doing something they believe in and nobody has objected, and finally, in some areas, a small group of people have had private discussions before doing something. The truth is that for a casual observer (on a user mailing list) it has been hard to understand how (and why) iptel.org projects move forward. It is also easy to get to the wrong conclusions and iptel.org has suffered from this lack of transparency. Over the last year we have worked to make such processes more transparent and predictable. We have focused on development and development decisions and have tried to encourage discussions on the mailing lists, as well as to create wishlists and (starting) roadmaps (see iptel.org website for these). However, we have not addressed management issues like how to solve conflicts where no consensus can be reached and so on. Jan (Janak) raised this issue during the Prague Developer Workshop where 18 people participated. I will below try to summarize what we had agreement on and what was decided. In order to make sure this is accurate, several participants with different views have reviewed this post before sending it on the mailing lists. (enclosed in objective tags) <objective> Principles agreed to (but up for discussion in the spirit of IETF): 1. We want discussions on the mailing lists and consensus when making decisions 2. We don't want a lot of bureaucracy that sounds nice, but that we don't need. This includes how to resolve all sorts of issues 3. In general, we trust people (that is individuals, not companies) who are most knowledgeable and most involved to make good decisions on our behalf. Hence, contributors should have more to say in the areas they contribute (ex. module developers) 4. We want to avoid "do-nothing" decisions, for example if everybody agrees, but one person says no, or people are split in two camps Decisions made (but up for discussion on the mailing lists in the spirit of IETF): * Discussions should be held on the developer mailing lists and no formal voting process should be used * If discussions just continue and no consensus can be reached, we want a small technical board (TB) to have the authority to make the decision on behalf of the community * This board should be elected by the community in an open election process * The technical board (TB) should also have the authority to decide how to resolve issues if there are no obvious precedence from earlier cases * The TB should be elected by the community for a pre-defined period * The TB should focus on issues related to day-to-day development of the projects, but should not manage on a day-to-day basis, just convene if there are issues that could not be resolved by consensus </objective> I try here to identify open issues not discussed/decided: * Should there be one TB for all iptel.org projects or one TB for each? * What should be the criteria for selecting people on the TB (if any)? * Who have voting rights when we vote on candidates for the TB? * What should be the term for a TB member? * Should this TB handle any issues beyond development? (ex. website content, iptel.org in the wider SIP context, relationship with other projects, packaging, longer-term positioning of projects, longer-term development goals, and so on) * If not, do we need another group that could handle such things? Best regards, Greger

9 10

Re: [Users] memory issues
by Daniel-Constantin Mierla 23 Apr '07

23 Apr '07

please keep cc-ing to mailing list so other users can get the response in case they have same questions, avoiding waste of time to solve same issue. Thanks, Daniel On 04/23/07 18:46, Tim Madorma wrote: > Great - thanks Daniel! > > On 4/23/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello Tim, >> >> On 04/23/07 18:36, Tim Madorma wrote: >> > Hey Daniel, >> > >> > When I looked at the ChangeLog, it does not seem to indicate that the >> > fix is there. >> > >> > >> http://openser.svn.sourceforge.net/svnroot/openser/branches/1.2/ChangeLog >> >> ChangeLog is usually left behind and updated just before the release. >> > >> > Should I be concerned about this? >> No >> > Should I just use the daily snapshot >> > instead? How stable is the daily snapshot? >> Daily snapshot is more or less what SVN 1.2 branch shows at the moment >> when the snapshot is done. So using SVN will guarantee the access to >> latest stable code. Snapshots and SVN of branch 1.2 should be the most >> stable in the 1.2.x release. There are no other commits to branch 1.2 >> but bug fixes. >> >> We publish daily snapshots just as backup when SVN is not available, >> otherwise SVN is advisable to use if you can use subversion. >> >> Cheers, >> Daniel >> > >> > Tim >> > >> > >> > On 4/23/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >> Hello, >> >> >> >> daily snapshots are enabled now for 1.2.x as well: >> >> >> >> http://www.openser.org/downloads/snapshots/openser-1.2.x/ >> >> >> >> Cheers, >> >> Daniel >> >> >> >> On 04/23/07 17:47, Ovidiu Sas wrote: >> >> > Hi Tim, >> >> > >> >> > Check the download page from openser website: >> >> > http://www.openser.org/mos/view/Download/: >> >> > >> >> > The command that you need to run: >> >> > svn co >> http://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 >> >> > openser >> >> > >> >> > Make sure that you have svn installed. >> >> > >> >> > >> >> > Regards, >> >> > Ovidiu Sas >> >> > >> >> > On 4/23/07, Tim Madorma <tmadorma(a)gmail.com> wrote: >> >> >> Hi Daniel, >> >> >> >> >> >> I have run into a leak in 1.2 and I assume it is the same one that >> >> >> Ovidiu ran into. I see in your response that it was "backported to >> >> >> 1.2", but I'm not sure how to get the fix. When I look at the SVN >> >> >> repository at: >> >> >> http://www.openser.org/pub/openser/latest-1.2.x/, the date is >> earlier >> >> >> than the date of your email exchange so I don't think the fix has >> >> been >> >> >> added there. Can you please let me know how I can get it? >> >> >> >> >> >> thanks, >> >> >> Tim >> >> >> >> >> >> On 3/23/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> >> wrote: >> >> >> > Hello Ovidiu, >> >> >> > >> >> >> > On 03/23/07 17:04, Ovidiu Sas wrote: >> >> >> > > Hi Daniel, >> >> >> > > >> >> >> > > Can we backport this one to 1.2? >> >> >> > already done, two minutes after the commit in trunk. >> >> >> > >> >> >> > Cheers, >> >> >> > Daniel >> >> >> > >> >> >> > > >> >> >> > > >> >> >> > > Regards, >> >> >> > > Ovidiu Sas >> >> >> > > >> >> >> > > On 3/22/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> >> >> wrote: >> >> >> > >> Hello, >> >> >> > >> >> >> >> > >> the supposed fragmentation turned out to be a mem leak in >> pkg. >> >> >> Please >> >> >> > >> take the latest SVN version and try again to see if you >> got same >> >> >> > >> results. >> >> >> > >> >> >> >> > >> Thanks, >> >> >> > >> Daniel >> >> >> > >> >> >> >> > >> On 03/19/07 18:52, Christian Schlatter wrote: >> >> >> > >> > ... >> >> >> > >> >>> The memory statistics indeed show a high number of memory >> >> >> fragments: >> >> >> > >> >>> >> >> >> > >> >>> before 'out of memory': >> >> >> > >> >>> >> >> >> > >> >>> shmem:total_size = 536870912 >> >> >> > >> >>> shmem:used_size = 59607040 >> >> >> > >> >>> shmem:real_used_size = 60106488 >> >> >> > >> >>> shmem:max_used_size = 68261536 >> >> >> > >> >>> shmem:free_size = 476764424 >> >> >> > >> >>> shmem:fragments = 9897 >> >> >> > >> >>> >> >> >> > >> >>> after 'out of memory' (about 8000 calls per process): >> >> >> > >> >>> >> >> >> > >> >>> shmem:total_size = 536870912 >> >> >> > >> >>> shmem:used_size = 4171160 >> >> >> > >> >>> shmem:real_used_size = 4670744 >> >> >> > >> >>> shmem:max_used_size = 68261536 >> >> >> > >> >>> shmem:free_size = 532200168 >> >> >> > >> >>> shmem:fragments = 57902 >> >> >> > >> >>> >> >> >> > >> >>>> >> >> >> > >> >>>> You can try to compile openser with -DQM_JOIN_FREE >> (add it >> >> >> in DEFS >> >> >> > >> >>>> variable of Makefile.defs) and test again. Free >> fragments >> >> >> should be >> >> >> > >> >>>> merged and fragmentation should not occur -- processing >> >> >> will be >> >> >> > >> >>>> slower. We will try for next release to provide a better >> >> >> solution >> >> >> > >> >>>> for that. >> >> >> > >> >>> >> >> >> > >> >>> Compiling openser with -DQM_JOIN_FREE did not help. >> I'm not >> >> >> sure how >> >> >> > >> >>> big of a problem this fragmentation issue is. >> >> >> > >> >> What is the number of fragments with QM_JOIN_FREE after >> >> >> flooding? >> >> >> > >> > >> >> >> > >> > The numbers included above are with QM_JOIN_FREE enabled. >> >> >> > >> > >> >> >> > >> >>> Do you think it would make sense to restart our >> production >> >> >> openser >> >> >> > >> >>> instances from time to time just to make sure they're not >> >> >> running >> >> >> > >> >>> into this memory fragmentation limits? >> >> >> > >> >> The issue will occur only when the call rate reaches the >> >> >> limits of >> >> >> > >> >> the proxy's memory. Otherwise the chunks are reused. >> >> >> Transactions and >> >> >> > >> >> avps are rounded up to be sure there will be minimized the >> >> >> number of >> >> >> > >> >> different sizes for memory chunks. It wasn't reported too >> >> often, >> >> >> > >> >> maybe that's why no big attention was paid to it. This >> memory >> >> >> system >> >> >> > >> >> is in place since the beginning of ser. Alternative is >> to use >> >> >> sysv >> >> >> > >> >> shared memory, but is much slower, along with libc private >> >> >> memory >> >> >> > >> >> manager. >> >> >> > >> > >> >> >> > >> > I've done some more testing and the same out-of-memory >> stuff >> >> >> happens >> >> >> > >> > when I run sipp with 10 calls per second only. I tested >> with >> >> >> > >> > 'children=1' and I only could get through about 8200 calls >> >> (again >> >> >> > >> > those 8000 calls / process). And this is with QM_JOIN_FREE >> >> >> enabled. >> >> >> > >> > >> >> >> > >> > Memory statistics: >> >> >> > >> > >> >> >> > >> > before: >> >> >> > >> > shmem:total_size = 536870912 >> >> >> > >> > shmem:used_size = 2311976 >> >> >> > >> > shmem:real_used_size = 2335720 >> >> >> > >> > shmem:max_used_size = 2465816 >> >> >> > >> > shmem:free_size = 534535192 >> >> >> > >> > shmem:fragments = 183 >> >> >> > >> > >> >> >> > >> > after: >> >> >> > >> > shmem:total_size = 536870912 >> >> >> > >> > shmem:used_size = 1853472 >> >> >> > >> > shmem:real_used_size = 1877224 >> >> >> > >> > shmem:max_used_size = 2465816 >> >> >> > >> > shmem:free_size = 534993688 >> >> >> > >> > shmem:fragments = 547 >> >> >> > >> > >> >> >> > >> > So I'm not sure if this is really a fragmentation issue. 10 >> >> >> cps surely >> >> >> > >> > doesn't reach the proxy's memory. >> >> >> > >> > >> >> >> > >> > Thoughts? >> >> >> > >> > >> >> >> > >> > Christian >> >> >> > >> > >> >> >> > >> > >> >> >> > >> > >> >> >> > >> >> Cheers, >> >> >> > >> >> Daniel >> >> >> > >> >> >> >> >> > >> >>> >> >> >> > >> >>> thanks, >> >> >> > >> >>> Christian >> >> >> > >> >>> >> >> >> > >> >>>> >> >> >> > >> >>>> Cheers, >> >> >> > >> >>>> Daniel >> >> >> > >> >>>> >> >> >> > >> >>>> On 03/18/07 01:21, Christian Schlatter wrote: >> >> >> > >> >>>>> Christian Schlatter wrote: >> >> >> > >> >>>>> ... >> >> >> > >> >>>>>> >> >> >> > >> >>>>>> I always had 768MB shared memory configured though, >> so I >> >> >> still >> >> >> > >> >>>>>> can't explain the memory allocation errors I got. Some >> >> >> more test >> >> >> > >> >>>>>> runs revealed that I only get these errors when using >> >> a more >> >> >> > >> >>>>>> production oriented config that loads more modules >> than >> >> >> the one >> >> >> > >> >>>>>> posted in my earlier email. I now try to figure out >> what >> >> >> exactly >> >> >> > >> >>>>>> causes these memory allocation errors that happen >> >> >> reproducibly >> >> >> > >> >>>>>> after about 220s at 400 cps. >> >> >> > >> >>>>> >> >> >> > >> >>>>> I think I found the cause for the memory allocation >> >> >> errors. As >> >> >> > >> >>>>> soon as I include an AVP write operation in the routing >> >> >> script, I >> >> >> > >> >>>>> get 'out of memory' messages after a certain number of >> >> calls >> >> >> > >> >>>>> generated with sipp. >> >> >> > >> >>>>> >> >> >> > >> >>>>> The routing script to reproduce this behavior looks >> like >> >> >> (full >> >> >> > >> >>>>> config available at >> >> >> > >> >>>>> http://www.unc.edu/~cschlatt/openser/openser.cfg): >> >> >> > >> >>>>> >> >> >> > >> >>>>> route{ >> >> >> > >> >>>>> $avp(s:ct) = $ct; # commenting this line solves >> >> >> > >> >>>>> # the memory problem >> >> >> > >> >>>>> >> >> >> > >> >>>>> if (!method=="REGISTER") record_route(); >> >> >> > >> >>>>> if (loose_route()) route(1); >> >> >> > >> >>>>> >> >> >> > >> >>>>> if (uri==myself) rewritehost("xx.xx.xx.xx"); >> >> >> > >> >>>>> route(1); >> >> >> > >> >>>>> } >> >> >> > >> >>>>> >> >> >> > >> >>>>> route[1] { >> >> >> > >> >>>>> if (!t_relay()) sl_reply_error(); >> >> >> > >> >>>>> exit; >> >> >> > >> >>>>> } >> >> >> > >> >>>>> >> >> >> > >> >>>>> An example log file showing the 'out of memory' >> >> messages is >> >> >> > >> >>>>> available at >> >> >> http://www.unc.edu/~cschlatt/openser/openser.log . >> >> >> > >> >>>>> >> >> >> > >> >>>>> Some observations: >> >> >> > >> >>>>> >> >> >> > >> >>>>> - The 'out of memory' messages always appear after >> about >> >> >> 8000 test >> >> >> > >> >>>>> calls per worker process. One call consists of two SIP >> >> >> > >> >>>>> transactions and six end-to-end SIP messages. An >> openser >> >> >> with 8 >> >> >> > >> >>>>> children handles about 64'000 calls, whereas 4 children >> >> only >> >> >> > >> >>>>> handle about 32'000 calls. The sipp call rate doesn't >> >> >> matter, only >> >> >> > >> >>>>> number of calls. >> >> >> > >> >>>>> >> >> >> > >> >>>>> - The 8000 calls per worker process are independent >> >> from the >> >> >> > >> >>>>> amount of shared memory available. Running openser >> with -m >> >> >> 128 or >> >> >> > >> >>>>> -m 768 does not make a difference. >> >> >> > >> >>>>> >> >> >> > >> >>>>> - The more AVP writes are done in the script, the less >> >> >> calls go >> >> >> > >> >>>>> through. It looks like each AVP write is leaking memory >> >> >> (unnoticed >> >> >> > >> >>>>> by the memory statistics). >> >> >> > >> >>>>> >> >> >> > >> >>>>> - The fifo memory statistics do not reflect the 'out of >> >> >> memory' >> >> >> > >> >>>>> syslog messages. Even if openser does not route a >> >> single SIP >> >> >> > >> >>>>> message because of memory issues, the statistics still >> >> >> show a lot >> >> >> > >> >>>>> of 'free' memory. >> >> >> > >> >>>>> >> >> >> > >> >>>>> >> >> >> > >> >>>>> All tests were done with openser SVN 1.2 branch on >> Ubuntu >> >> >> dapper >> >> >> > >> >>>>> x86. I think the same is true for 1.1 version but I >> >> >> haven't tested >> >> >> > >> >>>>> that yet. >> >> >> > >> >>>>> >> >> >> > >> >>>>> >> >> >> > >> >>>>> Christian >> >> >> > >> >>>>> >> >> >> > >> >>> >> >> >> > >> >>> >> >> >> > >> > >> >> >> > >> > >> >> >> > >> > _______________________________________________ >> >> >> > >> > Users mailing list >> >> >> > >> > Users(a)openser.org >> >> >> > >> > http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >> > >> >> >> > >> >> >> >> > >> _______________________________________________ >> >> >> > >> Users mailing list >> >> >> > >> Users(a)openser.org >> >> >> > >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >> >> >> >> > > >> >> >> > >> >> >> > _______________________________________________ >> >> >> > Users mailing list >> >> >> > Users(a)openser.org >> >> >> > http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >> >> >> >> >> > >> >> >> > >> >

1 0

Re: [Users] memory issues
by Daniel-Constantin Mierla 23 Apr '07

23 Apr '07

Hello Tim, On 04/23/07 18:36, Tim Madorma wrote: > Hey Daniel, > > When I looked at the ChangeLog, it does not seem to indicate that the > fix is there. > > http://openser.svn.sourceforge.net/svnroot/openser/branches/1.2/ChangeLog ChangeLog is usually left behind and updated just before the release. > > Should I be concerned about this? No > Should I just use the daily snapshot > instead? How stable is the daily snapshot? Daily snapshot is more or less what SVN 1.2 branch shows at the moment when the snapshot is done. So using SVN will guarantee the access to latest stable code. Snapshots and SVN of branch 1.2 should be the most stable in the 1.2.x release. There are no other commits to branch 1.2 but bug fixes. We publish daily snapshots just as backup when SVN is not available, otherwise SVN is advisable to use if you can use subversion. Cheers, Daniel > > Tim > > > On 4/23/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> daily snapshots are enabled now for 1.2.x as well: >> >> http://www.openser.org/downloads/snapshots/openser-1.2.x/ >> >> Cheers, >> Daniel >> >> On 04/23/07 17:47, Ovidiu Sas wrote: >> > Hi Tim, >> > >> > Check the download page from openser website: >> > http://www.openser.org/mos/view/Download/: >> > >> > The command that you need to run: >> > svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 >> > openser >> > >> > Make sure that you have svn installed. >> > >> > >> > Regards, >> > Ovidiu Sas >> > >> > On 4/23/07, Tim Madorma <tmadorma(a)gmail.com> wrote: >> >> Hi Daniel, >> >> >> >> I have run into a leak in 1.2 and I assume it is the same one that >> >> Ovidiu ran into. I see in your response that it was "backported to >> >> 1.2", but I'm not sure how to get the fix. When I look at the SVN >> >> repository at: >> >> http://www.openser.org/pub/openser/latest-1.2.x/, the date is earlier >> >> than the date of your email exchange so I don't think the fix has >> been >> >> added there. Can you please let me know how I can get it? >> >> >> >> thanks, >> >> Tim >> >> >> >> On 3/23/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >> > Hello Ovidiu, >> >> > >> >> > On 03/23/07 17:04, Ovidiu Sas wrote: >> >> > > Hi Daniel, >> >> > > >> >> > > Can we backport this one to 1.2? >> >> > already done, two minutes after the commit in trunk. >> >> > >> >> > Cheers, >> >> > Daniel >> >> > >> >> > > >> >> > > >> >> > > Regards, >> >> > > Ovidiu Sas >> >> > > >> >> > > On 3/22/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> >> wrote: >> >> > >> Hello, >> >> > >> >> >> > >> the supposed fragmentation turned out to be a mem leak in pkg. >> >> Please >> >> > >> take the latest SVN version and try again to see if you got same >> >> > >> results. >> >> > >> >> >> > >> Thanks, >> >> > >> Daniel >> >> > >> >> >> > >> On 03/19/07 18:52, Christian Schlatter wrote: >> >> > >> > ... >> >> > >> >>> The memory statistics indeed show a high number of memory >> >> fragments: >> >> > >> >>> >> >> > >> >>> before 'out of memory': >> >> > >> >>> >> >> > >> >>> shmem:total_size = 536870912 >> >> > >> >>> shmem:used_size = 59607040 >> >> > >> >>> shmem:real_used_size = 60106488 >> >> > >> >>> shmem:max_used_size = 68261536 >> >> > >> >>> shmem:free_size = 476764424 >> >> > >> >>> shmem:fragments = 9897 >> >> > >> >>> >> >> > >> >>> after 'out of memory' (about 8000 calls per process): >> >> > >> >>> >> >> > >> >>> shmem:total_size = 536870912 >> >> > >> >>> shmem:used_size = 4171160 >> >> > >> >>> shmem:real_used_size = 4670744 >> >> > >> >>> shmem:max_used_size = 68261536 >> >> > >> >>> shmem:free_size = 532200168 >> >> > >> >>> shmem:fragments = 57902 >> >> > >> >>> >> >> > >> >>>> >> >> > >> >>>> You can try to compile openser with -DQM_JOIN_FREE (add it >> >> in DEFS >> >> > >> >>>> variable of Makefile.defs) and test again. Free fragments >> >> should be >> >> > >> >>>> merged and fragmentation should not occur -- processing >> >> will be >> >> > >> >>>> slower. We will try for next release to provide a better >> >> solution >> >> > >> >>>> for that. >> >> > >> >>> >> >> > >> >>> Compiling openser with -DQM_JOIN_FREE did not help. I'm not >> >> sure how >> >> > >> >>> big of a problem this fragmentation issue is. >> >> > >> >> What is the number of fragments with QM_JOIN_FREE after >> >> flooding? >> >> > >> > >> >> > >> > The numbers included above are with QM_JOIN_FREE enabled. >> >> > >> > >> >> > >> >>> Do you think it would make sense to restart our production >> >> openser >> >> > >> >>> instances from time to time just to make sure they're not >> >> running >> >> > >> >>> into this memory fragmentation limits? >> >> > >> >> The issue will occur only when the call rate reaches the >> >> limits of >> >> > >> >> the proxy's memory. Otherwise the chunks are reused. >> >> Transactions and >> >> > >> >> avps are rounded up to be sure there will be minimized the >> >> number of >> >> > >> >> different sizes for memory chunks. It wasn't reported too >> often, >> >> > >> >> maybe that's why no big attention was paid to it. This memory >> >> system >> >> > >> >> is in place since the beginning of ser. Alternative is to use >> >> sysv >> >> > >> >> shared memory, but is much slower, along with libc private >> >> memory >> >> > >> >> manager. >> >> > >> > >> >> > >> > I've done some more testing and the same out-of-memory stuff >> >> happens >> >> > >> > when I run sipp with 10 calls per second only. I tested with >> >> > >> > 'children=1' and I only could get through about 8200 calls >> (again >> >> > >> > those 8000 calls / process). And this is with QM_JOIN_FREE >> >> enabled. >> >> > >> > >> >> > >> > Memory statistics: >> >> > >> > >> >> > >> > before: >> >> > >> > shmem:total_size = 536870912 >> >> > >> > shmem:used_size = 2311976 >> >> > >> > shmem:real_used_size = 2335720 >> >> > >> > shmem:max_used_size = 2465816 >> >> > >> > shmem:free_size = 534535192 >> >> > >> > shmem:fragments = 183 >> >> > >> > >> >> > >> > after: >> >> > >> > shmem:total_size = 536870912 >> >> > >> > shmem:used_size = 1853472 >> >> > >> > shmem:real_used_size = 1877224 >> >> > >> > shmem:max_used_size = 2465816 >> >> > >> > shmem:free_size = 534993688 >> >> > >> > shmem:fragments = 547 >> >> > >> > >> >> > >> > So I'm not sure if this is really a fragmentation issue. 10 >> >> cps surely >> >> > >> > doesn't reach the proxy's memory. >> >> > >> > >> >> > >> > Thoughts? >> >> > >> > >> >> > >> > Christian >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> >> Cheers, >> >> > >> >> Daniel >> >> > >> >> >> >> > >> >>> >> >> > >> >>> thanks, >> >> > >> >>> Christian >> >> > >> >>> >> >> > >> >>>> >> >> > >> >>>> Cheers, >> >> > >> >>>> Daniel >> >> > >> >>>> >> >> > >> >>>> On 03/18/07 01:21, Christian Schlatter wrote: >> >> > >> >>>>> Christian Schlatter wrote: >> >> > >> >>>>> ... >> >> > >> >>>>>> >> >> > >> >>>>>> I always had 768MB shared memory configured though, so I >> >> still >> >> > >> >>>>>> can't explain the memory allocation errors I got. Some >> >> more test >> >> > >> >>>>>> runs revealed that I only get these errors when using >> a more >> >> > >> >>>>>> production oriented config that loads more modules than >> >> the one >> >> > >> >>>>>> posted in my earlier email. I now try to figure out what >> >> exactly >> >> > >> >>>>>> causes these memory allocation errors that happen >> >> reproducibly >> >> > >> >>>>>> after about 220s at 400 cps. >> >> > >> >>>>> >> >> > >> >>>>> I think I found the cause for the memory allocation >> >> errors. As >> >> > >> >>>>> soon as I include an AVP write operation in the routing >> >> script, I >> >> > >> >>>>> get 'out of memory' messages after a certain number of >> calls >> >> > >> >>>>> generated with sipp. >> >> > >> >>>>> >> >> > >> >>>>> The routing script to reproduce this behavior looks like >> >> (full >> >> > >> >>>>> config available at >> >> > >> >>>>> http://www.unc.edu/~cschlatt/openser/openser.cfg): >> >> > >> >>>>> >> >> > >> >>>>> route{ >> >> > >> >>>>> $avp(s:ct) = $ct; # commenting this line solves >> >> > >> >>>>> # the memory problem >> >> > >> >>>>> >> >> > >> >>>>> if (!method=="REGISTER") record_route(); >> >> > >> >>>>> if (loose_route()) route(1); >> >> > >> >>>>> >> >> > >> >>>>> if (uri==myself) rewritehost("xx.xx.xx.xx"); >> >> > >> >>>>> route(1); >> >> > >> >>>>> } >> >> > >> >>>>> >> >> > >> >>>>> route[1] { >> >> > >> >>>>> if (!t_relay()) sl_reply_error(); >> >> > >> >>>>> exit; >> >> > >> >>>>> } >> >> > >> >>>>> >> >> > >> >>>>> An example log file showing the 'out of memory' >> messages is >> >> > >> >>>>> available at >> >> http://www.unc.edu/~cschlatt/openser/openser.log . >> >> > >> >>>>> >> >> > >> >>>>> Some observations: >> >> > >> >>>>> >> >> > >> >>>>> - The 'out of memory' messages always appear after about >> >> 8000 test >> >> > >> >>>>> calls per worker process. One call consists of two SIP >> >> > >> >>>>> transactions and six end-to-end SIP messages. An openser >> >> with 8 >> >> > >> >>>>> children handles about 64'000 calls, whereas 4 children >> only >> >> > >> >>>>> handle about 32'000 calls. The sipp call rate doesn't >> >> matter, only >> >> > >> >>>>> number of calls. >> >> > >> >>>>> >> >> > >> >>>>> - The 8000 calls per worker process are independent >> from the >> >> > >> >>>>> amount of shared memory available. Running openser with -m >> >> 128 or >> >> > >> >>>>> -m 768 does not make a difference. >> >> > >> >>>>> >> >> > >> >>>>> - The more AVP writes are done in the script, the less >> >> calls go >> >> > >> >>>>> through. It looks like each AVP write is leaking memory >> >> (unnoticed >> >> > >> >>>>> by the memory statistics). >> >> > >> >>>>> >> >> > >> >>>>> - The fifo memory statistics do not reflect the 'out of >> >> memory' >> >> > >> >>>>> syslog messages. Even if openser does not route a >> single SIP >> >> > >> >>>>> message because of memory issues, the statistics still >> >> show a lot >> >> > >> >>>>> of 'free' memory. >> >> > >> >>>>> >> >> > >> >>>>> >> >> > >> >>>>> All tests were done with openser SVN 1.2 branch on Ubuntu >> >> dapper >> >> > >> >>>>> x86. I think the same is true for 1.1 version but I >> >> haven't tested >> >> > >> >>>>> that yet. >> >> > >> >>>>> >> >> > >> >>>>> >> >> > >> >>>>> Christian >> >> > >> >>>>> >> >> > >> >>> >> >> > >> >>> >> >> > >> > >> >> > >> > >> >> > >> > _______________________________________________ >> >> > >> > Users mailing list >> >> > >> > Users(a)openser.org >> >> > >> > http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >> > >> >> > >> >> >> > >> _______________________________________________ >> >> > >> Users mailing list >> >> > >> Users(a)openser.org >> >> > >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >> >> >> > > >> >> > >> >> > _______________________________________________ >> >> > Users mailing list >> >> > Users(a)openser.org >> >> > http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >> >> >> > >> >

1 0

[Users] how to refresh REGISTER state when restarting redundant proxy
by Alan Crosswell 23 Apr '07

23 Apr '07

We are using t_replicate() to replicate REGISTERs among redundant proxies (and a separate presence server that needs REGISTER for pua_bla). Before reinventing the wheel here, I thought I'd ask if others already have a method in place to re-sync a restarted proxy's state? I guess one way is to pull the usrloc data from mysql.... Another would be to somehow ask the proxy to walk the usrloc table and do a bunch of t_registers()... Any thoughts appreciated. /a

9 25

[Users] question about ACK
by Papadopoulos Georgios 23 Apr '07

23 Apr '07

Hello all, In the following call to PSTN, Openser gets into an infinite loop when the ACK is received. It starts sending the ACK to itself. Is this because the ACK is not right or could it be a misconfiguration in my part? thank you for any help George # U 2007/04/18 13:44:46.759951 213.5.17.223:5058 -> 213.5.43.4:5060 INVITE sip:2116873831@sip.altecnet.gr:5060 SIP/2.0. Max-Forwards: 70. Content-Length: 378. Via: SIP/2.0/UDP 213.5.17.223:5058;branch=z9hG4bK08349b43d. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>. CSeq: 424044974 INVITE. Supported: 100rel. Supported: timer. Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE. Content-Type: application/sdp. Supported: replaces. Proxy-Authorization:Digest response="87bc0f48f292fa5213368434f8712c39",username="demo1",realm="alte cnet.gr",nonce="4625f7cabd05f0720edd9ab601211b1759066235",uri="sip:21168 73831@sip.altecnet.gr:5060". Contact: demo1 <sip:demo1@213.5.17.223:5058>. User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. . v=0. o=MxSIP 0 1153034350 IN IP4 213.5.17.223. s=SIP Call. c=IN IP4 213.5.17.223. t=0 0. m=audio 57994 RTP/AVP 8 0 97 98 109 99 4 18. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:97 G726-16/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:109 G726-32/8000. a=rtpmap:99 G726-40/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729/8000. a=ptime:20. a=sendrecv. a=silenceSupp:on - - - -. # U 2007/04/18 13:44:46.764816 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 213.5.17.223:5058;branch=z9hG4bK08349b43d;rport=5058. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>. CSeq: 424044974 INVITE. Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. . # U 2007/04/18 13:44:46.764967 213.5.43.4:5060 -> 213.5.1.6:5060 INVITE sip:400#2116873831@213.5.1.6:5060 SIP/2.0. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Max-Forwards: 69. Content-Length: 378. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>. CSeq: 424044974 INVITE. Supported: 100rel. Supported: timer. Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE. Content-Type: application/sdp. Supported: replaces. Contact: demo1 <sip:demo1@213.5.17.223:5058>. User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. . v=0. o=MxSIP 0 1153034350 IN IP4 213.5.17.223. s=SIP Call. c=IN IP4 213.5.17.223. t=0 0. m=audio 57994 RTP/AVP 8 0 97 98 109 99 4 18. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:97 G726-16/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:109 G726-32/8000. a=rtpmap:99 G726-40/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729/8000. a=ptime:20. a=sendrecv. a=silenceSupp:on - - - -. # U 2007/04/18 13:44:46.781896 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2007/04/18 13:44:46.913667 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:46.914048 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:47.413787 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:47.414278 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.413460 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.413952 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.582989 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.583304 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.718798 213.5.17.223:5058 -> 213.5.43.4:5060 ACK sip:2116873831@sip.altecnet.gr:5060 SIP/2.0. Max-Forwards: 70. Content-Length: 0. Via: SIP/2.0/UDP 213.5.17.223:5058;branch=z9hG4bKd1e7e81f7. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. CSeq: 424044974 ACK. Proxy-Authorization:Digest response="87bc0f48f292fa5213368434f8712c39",username="demo1",realm="alte cnet.gr",nonce="4625f7cabd05f0720edd9ab601211b1759066235",uri="sip:21168 73831@sip.altecnet.gr:5060". User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. . # U 2007/04/18 13:44:48.719303 213.5.43.4:5060 -> 213.5.43.4:5060 ACK sip:213.5.43.4;lr=on;ftag=299ad500d129e32 SIP/2.0. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Max-Forwards: 69. Content-Length: 0. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.2. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bKd1e7e81f7. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. CSeq: 424044974 ACK. Proxy-Authorization:Digest response="87bc0f48f292fa5213368434f8712c39",username="demo1",realm="alte cnet.gr",nonce="4625f7cabd05f0720edd9ab601211b1759066235",uri="sip:21168 73831@sip.altecnet.gr:5060". User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. P-hint: rr-enforced. . 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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Re: [Users] Re: Multiple invites
by venkatesh.d 23 Apr '07

23 Apr '07

hi ravi prakash, route[1] { append_branch("sip:C@xx:xx:xx:xx"); t_relay(); } above will work .but i have not enable nat routing in this .when i enabled nat routing with media proxy same problem i am facing raviprakash sunkara wrote: > Hi Klaus , > Thanks for replying , > > if(method=="INVITE" && uri=~"sip:B@xx.xxx.xx ") > { > setuser("C"); > append_branch() > t_relay(); > }; > > The B's Information is override by C , But Call goes to C , only, > > When A invites B, then B and C has ring parallel. > > On 4/23/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > Take a look at append_branch(). (core documentation) > > regards > klaus > > raviprakash sunkara wrote: > > Hello Users, > > > > For My Sip Servicing, There is Call Hunting Features > > > > This had done , When I forwarded the Asterisk Server. > > But I want to implements in OpenSER itself, > > > > When A Calls to B, the proxy has to send to invites to B and C, So that B > > and C rings simultaneously > > > > >

1 0

[Users] Multiple invites
by raviprakash sunkara 23 Apr '07

23 Apr '07

Hello Users, For My Sip Servicing, There is Call Hunting Features This had done , When I forwarded the Asterisk Server. But I want to implements in OpenSER itself, When A Calls to B, the proxy has to send to invites to B and C, So that B and C rings simultaneously -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

2 3

RE: [Serusers] Forward on busy hangs up actual call
by Rosa De Santis 23 Apr '07

23 Apr '07

Hi Greg, thanks for answering. I have tested with different UAs. I used XLite softphone and Grandstream hardphones, and always happens. Is there any configuration to avoid this? Thanks again. Rosa. ---------------------------------------- > Subject: Re: [Serusers] Forward on busy hangs up actual call > Date: Sat, 21 Apr 2007 07:56:30 +0200 > From: greger(a)teigre.com > To: rosadesantis(a)hotmail.com > CC: serusers(a)iptel.org > > Interesting... > However, this is a bug in A's UA. You should test on error codes though. > g-) > > ------- Original message ------- > From: Rosa De Santis > Sent: 20.4.'07, 22:26 > > > > > Hi all. > > Please, I have a problem forwarding to voicemail when busy. > > I have this in my failure route to forward to voicemail server: > > > > failure_route[1] { > > revert_uri(); > > rewritehostport("xx.xxx.xxx.xxx:5070"); > > append_branch(); > > t_relay(); > > break; > > } > > > > When A is busy talking with B, and C calls A then C is forwarded to voicemail. The forwarding works fine, but the call between A and B ends. > > > > Please, what am I doing wrong? > > Any help ¿? > > > > Thanks, Rosa. > > > > _________________________________________________________________ > > Descubre Live.com - tu propia página de inicio, personalizada para ver rápidamente todo lo que te interesa en un mismo sitio. > > http://www.live.com/getstarted_____________________________________________… > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > _________________________________________________________________ Descubre Live.com - tu propia página de inicio, personalizada para ver rápidamente todo lo que te interesa en un mismo sitio. http://www.live.com/getstarted

1 0

[Users] new developer: Henning Westerholt
by Daniel-Constantin Mierla 23 Apr '07

23 Apr '07

Hello, I want to welcome a new developer of OpenSER: Henning Westerholt, representing 1und1, Germany. He has been very active contributor with patches in the last time and now he just added a new module: cfgutils -- to become the silo of small and useful configuration script utilities. Cheers, Daniel

1 0

[Users] new module: cfgutils
by Henning Westerholt 23 Apr '07

23 Apr '07

Hello all, i've added a new module into trunk, called "cfgutils". It provides functions for random number generation, to introduce execution delay of the server and make random decisions based on a probability value. The sleep functions can be used to work around buggy user agents or gateways, the random functions are useful for load shedding on server overload. For usage and configuration of this module please refer to the README file. Credits for the sleep functions belongs to Carsten Bock, BASIS AudioNet GmbH. Best regards, Henning Westerholt

1 0

[Serusers] SER Running but .......
by micadeyeye＠crg.ee.uct.ac.za 23 Apr '07

23 Apr '07

Hi, I already have SER running on my PC but I couldn't go further to establish a communication between two sip phones (softphones on my system). I tried using EKIGA softphone and LINPHONE in UBUNTU but futile. Questions: 1. Can I use SER without specifying "LISTEN=XX.XX.XX.XX" because it was not included in the ser.cfg file which i recently downloaded? 2. Do I need to get NAT transversal inplace before communication can be established despite the use of 2 softphone on my PC? 3. I did the required modification to ser.cfg in order to use Mediaproxy but I keep getting "1 error" each time I restart SER. The modification made was this: modparam("mediaproxy", "mediaproxy-socket", "/var/run/mediaproxy.sock") changed to modparam("mediaproxy", "mediaproxy-select", "/var/run/proxydispatcher.sock") KINDLY HELP OUT. MIKE

1 0

[Users] transparent SIP proxy behind NAT
by Jerome Drouet 23 Apr '07

23 Apr '07

Hello, here is my network architecture : user (private IP ) ------- NAT(user) ------- (INTERNET) ------- SIPserver | | NAT(OpenSER) | | OpenSER transparent proxy or in a few words : - user is behind a NAT - OpenSER proxy is behind another NAT (public IP of the NAT : 82.x.x.x , private IP of the OpenSER proxy : 192.168.y.y) - the SIP server has a public IP address (51.z.z.z) The user NAT redirects traffic destinated to SIPserver (protocol UDP, port 5060) to the public IP of the OpenSER NAT, which of course redirects it to the private OpenSER proxy address I'd like to configure OpenSER as a transparent proxy server, just adding a field in the header of register messages during the forward to the SIPserver : so here are the important parts of the openser.cfg : **************************************** listen=192.168.y.y:5060 route { ... if (method=="REGISTER") { append_hf("Field : value"); route(2) return; } ... } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler if(!t_relay("udp:51.z.z.z:5060")) { sl_reply_error(); }; exit; } *************************************** unfortunatly, the "Via" field added by OpenSER contains its private address : i'd like OpenSER to use the IP address of its NAT instead : Via: SIP/2.0/UDP 192.168.y.y;.... ---> Via: SIP/2.0/UDP 82.x.x.x:5060;..... so that the SIPserver will respond through my OpenSER proxy to the user is there any way to do this ? it that can help, i'm running openser 1.2.0 from debian packages available from openser.org in the case openSER cannot do this, do you know any other software i don't know which could do the same things ? Regards, Jérôme

2 1

[Serusers] auth_identity module
by Tuan Nguyen 23 Apr '07

23 Apr '07

hi, did someone check auth_identity module. I have a problem with it: in ser configuration file, for authorizer service, need 4 parameters: # -- Authorizer Service parameters ---------- #-- privatekey_path ------------------------ modparam("auth_identity","privatekey_path",".../etc/CertAuth/private/m1key.pem") #--- certificate_path ----------------------- modparam("auth_identity","certificate_path","var/www/ssl/m1cert.crt") #--- certificate_url ------------------------ modparam("auth_identity","certificate_url","http://..../m1cert.crt") #--- msg_timeout ---------------------------- modparam("auth_identity","msg_timeout",600) when i run it, there is error: WARNING: no fork mode 0(9496) init_tcp: using epoll_lt as the io watch method (auto detected) 0(9496) Maxfwd module- initializing 0(9496) AUTH_INDENTITY:mod_init: unable to open certificate 'No such file or directory' 0(9496) init_mod(): Error while initializing module auth_identity ERROR: error while initializing modules I have checked the path of this certificate (that I had created): mml@kaneway:/var/www/ssl$ ls m1cert.crt I don't know why unable to open certificate with 'No such file or directory'. Can someone tell me what is my problem here. Thanks in advance Tuan.

3 2

[Serusers] Forward on busy hangs up actual call
by Rosa De Santis 21 Apr '07

21 Apr '07

Hi all. Please, I have a problem forwarding to voicemail when busy. I have this in my failure route to forward to voicemail server: failure_route[1] { revert_uri(); rewritehostport("xx.xxx.xxx.xxx:5070"); append_branch(); t_relay(); break; } When A is busy talking with B, and C calls A then C is forwarded to voicemail. The forwarding works fine, but the call between A and B ends. Please, what am I doing wrong? Any help ¿? Thanks, Rosa. _________________________________________________________________ Descubre Live.com - tu propia página de inicio, personalizada para ver rápidamente todo lo que te interesa en un mismo sitio. http://www.live.com/getstarted

2 1

[Serusers] MailboxURL.py
by Meidiana Ten 20 Apr '07

20 Apr '07

Hello all, Thanx for the answers. I have my SEMS IVR run now. But I have new problems here. ;b I can't connect to IMAP inbox as written on script below: avp_write("Mailbox-URL=imap://user:password@imapserver:143/INBOX", "$mailbox_url"); Then I try to install courier-imap as IMAP server to store the voicemail and configure MailboxURL.py to match the URL. Still can't work though. I didn't know what I was configured there. Please some advise. Thanx Regards, Meidiana --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

1 0

Re: [Serusers] Help needed!
by Libor Chocholaty 20 Apr '07

20 Apr '07

I am sorry I do not know this Redhat package. However as I remember you can use following: $ make prefix=/opt/ser modules $ make prefix=/opt/ser $ su # make prefix=/opt/ser install All you need to compile and install SER is in INSTALL file. You will try cvs version of SER 2.0 also. I tried it on Fedora Core 5 and it works well for my simple test. You probably will have to install several development packages. One more think is good to remember -- do not use CVS trunk in these days and use tag rel_2_0_0 instead. So checkout ser this way: $ cvs co -r rel_2_0_0 sip_router Libor Mukesh Prajapati wrote: > thnks for your response. > I have downloaded SER setup from http://ftp.iptel.org/pub/ser/0.9.6 > it is ser-0.9.6_linux_i386.tar.gz > > i installed it on RadHat system, so in current directory usr is > created. i found binaries at usr/local/sbin directory. > > now the problem is i can't start the ser server. > so, would u please help me in this regards. > > thanks > Mukesh > > */Libor Chocholaty <libor_ml1(a)mts.cz>/* wrote: > > Mukesh Prajapati wrote: > > Hi all, > > I am new to SER > > I m trying to install n start SER on RadHat Linux. So, please > guide me > > how to find SER source n install. > > n how can I start SER. > > waiting for reply. > $ export CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/ser > $ cvs login > Logging in to :pserver:anonymous@cvs.berlios.de:2401/cvsroot/ser > CVS password: <<-- do not pass any password, only press enter -->> > $ cvs co sip_router > ---snip--- > $ cd sip_router > $ less INSTALL > > Libor > > > ------------------------------------------------------------------------ > Ahhh...imagining that irresistible "new car" smell? > Check out new cars at Yahoo! Autos. > <http://us.rd.yahoo.com/evt=48245/*http://autos.yahoo.com/new_cars.html;_ylc…>

1 0

[Users] Proxy Route Processing
by Victor Gamov 20 Apr '07

20 Apr '07

Hi guys! I have a question about "Proxy Route Processing" I have following request comes to my OpenSER (xxx.xxx.xxx.xxx is IP of this OpenSER): PRACK sip:1111@xxx.xxx.xxx.xxx:5060;nat=yes SIP/2.0 Via: SIP/2.0/UDP 192.168.100.77:5060;branch=z9hG4bK_00179A52BC45_T59048F1A From: "Vit-Horosho" <sip:vit-horosho@xxx.xxx.xxx.xxx>;tag=00179A52BC45_T1304371222 To: <sip:1111@xxx.xxx.xxx.xxx:5060>;tag =0015E9A0D47D_T1002915332 Call-ID: CALL_ID3_00179A52BC45_T2098965184(a)192.168.100.77 RAck: 1909013112 135588358 INVITE CSeq: 135588359 PRACK Contact: <sip:vit-horosho@192.168.100.77:5060> Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=00179A52BC45_T1304371222> Content-Length: 0 As we reading in RFC3261 "16.12 Summary of Proxy Route Processing" part: 1. Request-URI must be rewrited based on location service 2. Route header must be removed because it indicate this proxy 3. Proxy will forward the request to the resource indicated by the URI in the topmost Route header field value (but we have not this Route after step 2) or in the Request-URI if no Route header field is present (so it must be forwarded to URI getted on step 1) But when I inspect OpenSER debug messages I found the following: 0(99796) parse_headers: flags=200 0(99796) is_preloaded: No 0(99796) grep_sock_info - checking if host==us: 11==11 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(99796) grep_sock_info - checking if port 5060 matches port 5060 0(99796) after_strict: Next hop: 'sip:xxx.xxx.xxx.xxx;lr=on;ftag=00179A52BC45_T1304371222' is loose router 0(99796) parse_headers: flags=ffffffffffffffff 0(99796) rewrite_uri: Rewriting Request-URI with 'sip:xxx.xxx.xxx;lr=on;ftag=00179A52BC45_T1304371222' 0(99796) after_strict: The last route URI: 'sip:xxx.xxx.xxx.xxx;lr=on;ftag=00179A52BC45_T1304371222' 0(99796) parse_headers: flags=ffffffffffffffff So request is not forwarded as I expect Is it my misconfiguration or my misunderstood of RFC3261 ? (the part of openser.cfg followed): ----------- cut here ---------------- # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); #xdbg("\n\troute[0]: before loose_route() ou=$ou; ru=$ru\n"); # 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); }; force_rport(); [ request processing with lookup() and actually route(1) followed ] ----------- /cut here ----------------

4 15

[Serusers] Help needed!
by Mukesh Prajapati 20 Apr '07

20 Apr '07

Hi all, I am new to SER I m trying to install n start SER on RadHat Linux. So, please guide me how to find SER source n install. n how can I start SER. waiting for reply. Thanks Mukesh --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

2 1

[Serusers] Re: [Serdev] Re: stateful forking?
by Cesc 20 Apr '07

20 Apr '07

Hi Atle! I knew it had to be sth with append_branch, but all (the few) examples I looked at had append_branch with a full sip uri as parameter ... I will try this, but I am sure this will work :) Cesc On 4/19/07, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > > Hi Cesc :) > > In this case I think I would write something like : > if(!message="REGISTER"){record_route()}; > > if(method=="MESSAGE"){ > t_on_reply("5"); > rewritehostport("SER3"); > append_branch(); > rewritehostport("SER2"); > t_relay(); > } > > - atle > > * Cesc <cesc.santa(a)gmail.com> [070419 19:26]: > > With some ser-ish routing code ... what i do now: > > > > if ( MESSAGE ) { > > t_on_reply("5"); > > rewritehostport( SER3); > > forward("IP_MCAST_LOC"); > > > > > > rewritehostport( SER2 ); > > route( ROUTE_RELAY_DO_TRELAY ); > > break; > > }; > > > > On 4/19/07, Cesc <cesc.santa(a)gmail.com> wrote: > > >Hi, > > > > > >Me = using ser 0.9.6 > > > > > >It is not very clear for me what I need to do, so bear with me :) > > > > > >I have a setup such: > > > > > >phone ..... SER1 ........ SER 2 > > > | ................. SER3 > > >So, one ser being frontend of the other 2 ... now, let say that phone > > >sends MESSAGE msgs, which I want both (ser 2 and 3 ) to receive them. > > >So, forking, right? > > >In a previous setup, it sort of work by doing t_newtran() and then > > >sequentially forward(ser2_ip) and t_forward_nonack(ser3_ip_port) ... > > >not done by me, the setup, i mean. But eventhough I set t_on_reply() > > >routes, I don't see the OK/errors msgs when coming back. > > >Why? > > >Even better, if I could set a different on_reply for ser2 and for ser3 ... > > > > > >Regards, > > > > > >Cesc > > > > > _______________________________________________ > > Serdev mailing list > > Serdev(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serdev >

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[Users] short remainder: IRC OpenSER meeting today 13:00 GMT
by Henning Westerholt 20 Apr '07

20 Apr '07

Hello all, just a short remainder about the IRC OpenSER meeting today at 13:00 GMT in the #openser channel on freenode.net. Information about this meeting: http://openser.org/dokuwiki/doku.php/development:irc-meeting-agenda-20-04-2… Please join us! -- Henning Westerholt

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Re: [Serusers] global contact list
by Vaclav Kubart 20 Apr '07

20 Apr '07

Hi, I'm very sorry for delayed reply but... ;-) Your config seems to be OK for first (and second) look. I have used list subscripriptions with EyeBeam in this way: I added the list URI in my contact list (exactly the same way like other contacts), so I had something like: contacts: - user1(a)somewhere.com - user2(a)somewhere.com - hr(a)somewhere.com - ... - franta-list(a)somewhere.com Where userx@... are 'normal' users and hr@... is resource list as defined in global resource list - in your case it could be bartaj-list(a)test.intra or hr2(a)test.intra. The franta-list@... is uri used to subscribe to user's resource list when I'm using my user with uri franta(a)somewhere.com (i.e. the resource list on XCAP is stored in user's directory as described in presence handbook). After the subscription is created, the list in contact window was expanded and seemd something like: - user1(a)somewhere.com - user2(a)somewhere.com - hr(a)somewhere.com | +- smith(a)somewhere.com +- ... - ... If I tried to use "Subscribe to contact list" feature, I think I had to put franta-list(a)somewhere.com as contact-list URI (or hr(a)somewhere.com could be used too). But in this case EyeBeam was not sending the Supported:eventlist header thus it was unusable for me... Have you seen any error messages in SER's log? Vaclav On Tue, Apr 17, 2007 at 05:01:24PM +0200, Barta János wrote: > Hello! > > Thanks for your answer again, and I'm sorry for the delayed answer, but I was out of office. > I attached my config file. > > My biggest problem is, that I don't know, what I should see :-(. > I have an eyeBeam window, that contains the "Calls & Conntats" parameters. > > In this window I can see my (user) contact list, but I didn't see the global one. > For example (the PC is my user contact list): > > -PC > -105 > -12345 > -barta > > and it is working fine, but I can not see here (in this window) the global one: > > sip:~# cat /var/www/xcap-root/rls-services/global/index > <?xml version="1.0" encoding="UTF-8"?> > <rls-services> > > <service uri="sip:bartaj-list@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > <service uri="sip:hr2@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > </rls-services> > > > I think in that case I should see the next tree: > > -PC > -105 > -12345 > -barta > > -Human resources > -bartaj > -12345 > > > > As You asked, I made a capture between the SER and XCAP. > > It was a little difficult, because I didn't see interesting traffic between the SER and XCAP, but a few minutes later I realised, that the SER for the communication (in that case) uses the loopback (lo) interface. > > You can see here the capture, that I made on the lo interface: > > 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [SYN] Seq=0 Len=0 MSS=16396 TSV=3808162073 TSER=0 WS=0 > 2 0.000056 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [SYN, ACK] Seq=0 Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162073 TSER=3808162073 WS=0 > 3 0.000065 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=1 Ack=1 Win=32767 Len=0 TSV=3808162073 TSER=3808162073 > 4 0.000072 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/pres-rules/users/12345/presence-rules.xml HTTP/1.1 > 5 0.000079 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [ACK] Seq=1 Ack=113 Win=32767 Len=0 TSV=3808162073 TSER=3808162073 > 6 0.000545 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK > 7 0.000559 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 Ack=262 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 > 8 0.000571 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic > 9 0.000578 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 Ack=870 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 > 10 0.005535 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [SYN] Seq=0 Len=0 MSS=16396 TSV=3808162079 TSER=0 WS=0 > 11 0.005556 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [SYN, ACK] Seq=0 Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162079 TSER=3808162079 WS=0 > 12 0.005565 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=1 Ack=1 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 13 0.005616 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/pres-rules/users/bartaj/presence-rules.xml HTTP/1.1 > 14 0.005623 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [ACK] Seq=1 Ack=114 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 15 0.005989 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK > 16 0.006000 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 Ack=262 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 17 0.006011 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic > 18 0.006017 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 Ack=870 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 > 19 2.979491 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/rls-services/global/index HTTP/1.1 > 20 2.979722 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK > 21 2.979734 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 Ack=1126 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 > 22 2.979745 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic > 23 2.979751 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 Ack=1887 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 > > > > You can see, that in the message number 19 the SER downloaded the index file. > > This file contains correct information: > > > sip:~# wget 127.0.0.1/xcap/rls-services/global/index > --17:22:43-- http://127.0.0.1/xcap/rls-services/global/index > => `index' > Connecting to 127.0.0.1:80... connected. > HTTP request sent, awaiting response... 200 OK > Length: 761 [text/plain] > > 100%[====================================>] 761 --.--K/s > > 17:22:43 (7.26 MB/s) - `index' saved [761/761] > > sip:~# > sip:~# cat index > <?xml version="1.0" encoding="UTF-8"?> > <rls-services> > > <service uri="sip:bartaj-list@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > <service uri="sip:hr2@test.intra"> > <list name="Human resources"> > <entry uri="sip:test1@test.intra"> > <display-name>bartaj</display-name> > </entry> > <entry uri="sip:test2@test.intra"> > <display-name>12345</display-name> > </entry> > </list> > <packages> > <package>presence</package> > </packages> > </service> > > </rls-services> > sip:~# > > > > And at last, how can I add query_rls_services and query_resource_list log informations into the config file? > > Thx, > Jani > > > > Hi, > this part of config file is not sufficient for me, but I guess that RLS related things stayed as in the original config you mentioned... In which way it doesn't work? (For example it returns non OK responses?) Please could you try to capture HTTP traffic between XCAP server and SER? (If SER really queries the right resource list document.) > > You can add more log information around important config parts (query_rls_services and query_resource_list functions are most important here). > > BTW the second message you sent is not subscription to resource list or user, it is subscription to event package 'sip-profile' (content of Event header field) which is not supported by SER. > > It is possible to use registration status as source for presence - in this case you have to have PA module on your server handling registrations and PA module parameter 'use_callbacks' must be set to 1. > > Vaclav > > > > -----Original Message----- > From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Sent: Tuesday, April 03, 2007 11:18 AM > To: Barta János > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] global contact list > > > _______________________________________________________________________________ > Állást keres? 3300+ lehetőséget kínál a CV-Online > Ne keressen tovább! Portálunkon Magyarország vezető vállalatainak több mint 3300 friss állásajánlatából válogathat! > > http://ad.adverticum.net/b/cl,1,6022,139845,209300/click.prm > sip:/opt/etc/ser# cat ./ser_pa.cfg > log_stderror=yes > debug=5 # debug level (cmd line: -dddddddddd) > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=2 > alias="test.intra" > #alias="ipp.test.intra" > alias="10.22.1.210" > mhomed=yes # usefull for multihomed hosts, small performance penalty > #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) > #tcp_poll_method="sigio_rt" > > # ------------------ module loading ---------------------------------- > > loadmodule "/opt/lib/ser/modules/sl.so" > loadmodule "/opt/lib/ser/modules/avp.so" > loadmodule "/opt/lib/ser/modules/avpops.so" > loadmodule "/opt/lib/ser/modules/tm.so" > loadmodule "/opt/lib/ser/modules/rr.so" > loadmodule "/opt/lib/ser/modules/maxfwd.so" > loadmodule "/opt/lib/ser/modules/usrloc.so" > loadmodule "/opt/lib/ser/modules/registrar.so" > loadmodule "/opt/lib/ser/modules/textops.so" > loadmodule "/opt/lib/ser/modules/mysql.so" > loadmodule "/opt/lib/ser/modules/dialog.so" > loadmodule "/opt/lib/ser/modules/rls.so" > loadmodule "/opt/lib/ser/modules/pa.so" > loadmodule "/opt/lib/ser/modules/presence_b2b.so" > loadmodule "/opt/lib/ser/modules/uri.so" > loadmodule "/opt/lib/ser/modules/uri_db.so" > loadmodule "/opt/lib/ser/modules/domain.so" > loadmodule "/opt/lib/ser/modules/fifo.so" > loadmodule "/opt/lib/ser/modules/xmlrpc.so" > loadmodule "/opt/lib/ser/modules/xlog.so" > #loadmodule "/opt/lib/ser/modules/unixsock.so" > loadmodule "/opt/lib/ser/modules/msilo.so" > > # Uncomment this if you want digest authentication > # mysql.so must be loaded ! > loadmodule "/opt/lib/ser/modules/auth.so" > #loadmodule "/opt/lib/ser/modules/auth_db.so" > > loadmodule "/opt/lib/ser/modules/acc_radius.so" > loadmodule "/opt/lib/ser/modules/mediaproxy.so" > loadmodule "/opt/lib/ser/modules/auth_radius.so" > > # ----------------- setting module-specific parameters --------------- > ############################# > modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") > modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") > > ######################### > # -- acc params -- # > ######################### > modparam("acc_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") > modparam("acc_radius", "log_flag", 1) > modparam("acc_radius", "log_missed_flag", 2) > > > modparam("msilo","registrar","sip:registrar@test.intra") > modparam("msilo","use_contact",0) > modparam("msilo","expire_time",7200) > > # -- usrloc params -- > > # -- auth params -- > # Uncomment if you are using auth module > # > #modparam("auth_db", "calculate_ha1", yes) > # > # If you set "calculate_ha1" parameter to yes (which true in this config), > # uncomment also the following parameter) > # > #modparam("auth_db", "password_column", "password") > > # -- rr params -- > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > > modparam("rls", "min_expiration", 200) > modparam("rls", "max_expiration", 300) > modparam("rls", "default_expiration", 300) > modparam("rls", "auth", "none") > modparam("rls", "xcap_root", "http://localhost/xcap") > modparam("rls", "reduce_xcap_needs", 1) > modparam("rls", "db_mode", 1) > modparam("rls", "db_url", "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > > modparam("pa", "use_db", 1) > # allow storing authorization requests for offline users into database > modparam("pa", "use_offline_winfo", 1) > # how often try to remove old stored authorization requests > modparam("pa", "offline_winfo_timer", 600) > # how long stored authorization requests live > modparam("pa", "offline_winfo_expiration", 600) > modparam("pa", "db_url", "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > # mode of PA authorization: none, implicit or xcap > modparam("pa", "auth", "xcap") > modparam("pa", "auth_xcap_root", "http://localhost/xcap") > # do not authorize watcherinfo subscriptions > modparam("pa", "winfo_auth", "none") > # use only published information if set to 0 > modparam("pa", "use_callbacks", 1) > # don't accept internal subscriptions from RLS, ... > modparam("pa", "accept_internal_subscriptions", 1) > # maximum value of Expires for subscriptions > modparam("pa", "max_subscription_expiration", 600) > # maximum value of Expires for publications > modparam("pa", "max_publish_expiration", 120) > # how often test if something changes and send NOTIFY > modparam("pa", "timer_interval", 10) > > # route for generated SUBSCRIBE requests for presence > modparam("presence_b2b", "presence_route", "<sip:127.0.0.1;transport=tcp;lr>") > # waiting time from error to new attepmt about SUBSCRIBE > modparam("presence_b2b", "on_error_retry_time", 60) > # how long wait for NOTIFY with Subscription-Status=terminated after unsubscribe > modparam("presence_b2b", "wait_for_term_notify", 33) > # how long before expiration send renewal SUBSCRIBE request > modparam("presence_b2b", "resubscribe_delta", 30) > # minimal time to send renewal SUBSCRIBE request from receiving previous response > modparam("presence_b2b", "min_resubscribe_time", 60) > # default expiration timeout > modparam("presence_b2b", "default_expiration", 3600) > # process internal subscriptions to presence events > modparam("presence_b2b", "handle_presence_subscriptions", 1) > > modparam("usrloc", "db_mode", 2) > modparam("domain", "db_mode", 1) > #modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > modparam("domain|uri_db|auth_db|usrloc|msilo", "db_url", "mysql://xxxxxxxxxxx:yyyyyyyyy@10.22.1.222/ser_pa") > > modparam("fifo", "fifo_file", "/tmp/ser_fifo") > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > # XML RPC > if (method == "POST" || method == "GET") { > create_via(); > dispatch_rpc(); > break; > } > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if (msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!method=="REGISTER") record_route(); > > if ((method=="INVITE") or (method=="BYE") or (method=="CANCEL")) > { > setflag(1); > } > > # 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 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 (!lookup_domain("To")) { > xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); > route(1); > break; > } > > if (method=="SUBSCRIBE") { > if (!t_newtran()) { > sl_reply_error(); > break; > }; > > if (@to.tag=="") { > # only for new subscriptions (with empty to tag) > > if (lookup_user("To")) { > # existing user -> it is subscription to PA > if (handle_subscription("registrar")) { > if ((@msg.event=~"presence\.winfo")) { > # new watcher info subscription > # sends one watcher info NOTIFY message with all saved authorization requests > xlog("L_ERR", "dumping stored winfo to %fu\n"); > dump_stored_winfo("registrar", "presence"); > } > else { > # new presence subscription > if ((@msg.event=~"presence") && (%subscription_status=="pending")) { > # if offline user and new pending subscription > if (!target_online("registrar")) { > #%subscription_status="waiting"; # store it as waiting subscription > xlog("L_ERR", "storing 'pending' winfo to: %tu, from: %fu\n"); > store_winfo("registrar"); > } > } > } > } > break; > } > > if ((@msg.supported=~"eventlist")) { > # such user doesn't exist and Supported header field > # -> probably RLS subscription > > if (lookup_domain("From")) { > if (lookup_user("From")) { > if (is_simple_rls_target("$uid-list")) { > # log(1, "it is simple subscription!\n"); > # takes From UID and makes XCAP query for user's > # list named "default" > if (!query_resource_list("default")) { > t_reply("404", "No such user list"); > break; > } > } > } > } > > if (!have_flat_list()) { > # query_resource_list failed or was not called > # do standard RLS query acording to To/AOR > if (!query_rls_services()) { > log(1, "XCAP query failed\n"); > t_reply("404", "No such list URI"); > break; > } > } > > > if (!radius_www_authorize("test.intra")) { > www_challenge("test.intra", "0"); > break; > }; > handle_rls_subscription("1"); > } > else { > # not resource list subscription -> invalid user > xlog("L_ERR", "subscription to invalid user %tu\n"); > t_reply("404", "User not found"); > } > > break; > } > else { > # renewal subscriptions - try to handle it as RLS and if failed, handle it as PA subscription > # FIXME: better will be test like existing_rls_subscription() > # and existing_subscription("registrar") > if (!handle_rls_subscription("0")) { > handle_subscription("registrar"); > } > break; > } > }; > > # get user (common for all other messages than SUBSCRIBE) > if (!lookup_user("To")) { > # log(1, "Unknown user - message should be forwarded?"); > # # break; > append_hf("P-hint: unknown user\r\n"); > route(1); > break; > } > > if (method=="PUBLISH") { > if (!t_newtran()) { > # log(1, "newtran error\n"); > sl_reply_error(); > break; > }; > handle_publish("registrar"); > > # deliver messages to online user > # TODO: only if user goes from offline to online? > if (target_online("registrar")) { > # log(1, "Dumping stored messages\n"); > # dump stored messages - route it through myself (otherwise routed via DNS!) > if (m_dump("sip:127.0.0.1")) { > xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); > } > } > > break; > }; > > if (method=="NOTIFY") { > if (!t_newtran()) { > log(1, "newtran error\n"); > sl_reply_error(); > break; > }; > # handle notification sent in internal subscriptions (presence_b2b) > if (!handle_notify()) { > t_reply("481", "Unable to handle notification"); > } > break; > }; > > if (method=="MESSAGE") { > > if (authorize_message("http://localhost/xcap")) { > > # use usrloc for delivery > if (lookup("location")) { > > log(1, "Delivering MESSAGE using usrloc\n"); > t_on_failure("1"); > if (!t_relay()) { > sl_reply_error(); > } > > break; > } > else { > # store messages for offline user > xlog("L_ERR", "MSILO: storing MESSAGE for %tu\n"); > > if (!t_newtran()) { > log(1, "newtran error\n"); > sl_reply_error(); > break; > }; > > # store only text messages NOT isComposing... ! > if (search("^(Content-Type|c):.*application/im-iscomposing\ xml.*")) { > log(1, "it is only isComposing message - ignored\n"); > t_reply("202", "Ignored"); > break; > } > > if (m_store("0", "sip:127.0.0.1")) { > # log(1, "MSILO: offline message stored\n"); > if (!t_reply("202", "Accepted")) { > sl_reply_error(); > }; > } else { > log(1, "MSILO: error storing offline message\n"); > if (!t_reply("503", "Service Unavailable")) { > sl_reply_error(); > }; > }; > break; > } > break; > } > else { > # log(1, "unauthorized message\n"); > sl_reply("403", "Forbidden"); > } > break; > } > > if (method=="REGISTER") { > > if (!radius_www_authorize("test.intra")) { > www_challenge("test.intra", "0"); > break; > }; > > save("location"); > > # dump stored messages - route it through myself (otherwise routed via DNS!) > if (m_dump("sip:127.0.0.1")) { > xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); > } > 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(); > # }; > #} > > > #failure_route[1] { > # # forwarding failed -- check if the request was a MESSAGE > # if (!method=="MESSAGE") { break; }; > # log(1, "MSILO: MESSAGE forward failed - storing it\n"); > # > # # we have changed the R-URI with the contact address, ignore it now > # if (m_store("0", "")) { > # t_reply("202", "Accepted"); > # } else { > # log(1, "MSILO: offline message NOT stored\n"); > # t_reply("503", "Service Unavailable"); > # }; > #} > > > > route[1] > { > > if (method=="INVITE") > { > t_on_reply("1"); > t_on_failure("1"); > use_media_proxy(); > } > else if (method=="BYE|CANCEL") { > end_media_session(); > } > > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > } > > > > onreply_route[1] > { > if (status=~"(183)|2[0-9][0-9]") { > use_media_proxy(); > }; > if (status=~"[3-6]0[0-9]") { > end_media_session(); > break; > } > } > failure_route[1] > { > # forwarding failed -- check if the request was a MESSAGE > if (!method=="MESSAGE") { break; }; > log(1, "MSILO: MESSAGE forward failed - storing it\n"); > # we have changed the R-URI with the contact address, ignore it now > if (m_store("0", "")) { > t_reply("202", "Accepted"); > } else { > log(1, "MSILO: offline message NOT stored\n"); > t_reply("503", "Service Unavailable"); > }; > end_media_session(); > }

1 0

[Serusers] Re: [Sems] IVR module can't be built
by Meidiana Ten 20 Apr '07

20 Apr '07

Hi Sumin, Thanx for replying my mail. I failed to compile ivr module. btw I'm using sems.0.10.0-rc2 as media server. Below is the error log: python2.2 py_comp -l py &&\ cp py/*.pyc ../../core/lib Listing py ... make[1]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' g++ -MM IvrAudio.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > IvrAudio.d g++ -MM Ivr.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > Ivr.d g++ -MM IvrDialogBase.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > IvrDialogBase.d g++ -MM IvrSipDialog.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > IvrSipDialog.d g++ -MM IvrSipReply.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > IvrSipReply.d g++ -MM IvrSipRequest.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > IvrSipRequest.d g++ -MM IvrUAC.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing > IvrUAC.d make[1]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' make[1]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing -c IvrAudio.cpp -o IvrAudio.o g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.2 -fno-strict-aliasing -c Ivr.cpp -o Ivr.o Ivr.cpp:54: 'PyGILState_STATE' is used as a type, but is not defined as a type. Ivr.cpp: In constructor `PythonGIL::PythonGIL()': Ivr.cpp:56: `gst' undeclared (first use this function) Ivr.cpp:56: (Each undeclared identifier is reported only once for each function it appears in.) Ivr.cpp:56: `PyGILState_Ensure' undeclared (first use this function) Ivr.cpp: In destructor `PythonGIL::~PythonGIL()': Ivr.cpp:57: `PyGILState_Release' undeclared (first use this function) Ivr.cpp: In member function `bool IvrFactory::loadScript(const std::string&)': Ivr.cpp:363: invalid conversion from `const char*' to `char*' Ivr.cpp: In member function `bool IvrDialog::callPyEventHandler(char*, char*, ...)': Ivr.cpp:640: `PyBool_Check' undeclared (first use this function) make[1]: *** [Ivr.o] Error 1 make[1]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' make: [all] Error 2 (ignored) echo "Installing Python object files..." Installing Python object files... for f in log.pyc ; do \ if [ -n "py/$f" ]; then \ touch /usr/local/lib/sems/plug-in//$f; \ install -m 755 py/$f /usr/local/lib/sems/plug-in//$f; \ fi ; \ done mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//ivr.so ; \ install -m 755 ivr.so /usr/local/lib/sems/plug-in/ install: cannot stat `ivr.so': No such file or directory make: *** [install] Error 1 I used "make install" to compile the module. I'd like to ask another question too...Where does SEMS voicemail usually store voice message? And what is the type of the message store (wav or text file)? Thanx before Regards, Meidiana seo <giganti71(a)gmail.com> wrote: Hi Meidiana, Do you mean that you failed to compile ivr module OR you failed to load ivr plug-in for mailbox application? If you failed to compile ivr module, could you send compile error log? If you didn't see any compile error , I think you have to check if there is ivr.so file on your system and then check if ivr.so file is in the plug-in path which is defined in sems.conf file. If you still have an issue even though all plug-in files are in the plug-in path, could you attach sems logs? Thanks. Sumin On 4/17/07, Meidiana Ten <gotcha_em(a)yahoo.com> wrote: Hi there, Thanx before for my past questions. Now, I want to add mailbox application on my server but I cannot find IVR plugin Then, I read some tutorial that I must make the plugin itself I've followed the step but it failed. What should I do then? Please advise Thanx Regards, Meidiana --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check out new cars at Yahoo! Autos. _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

5 6

[Serusers] fail to store voicemail
by Meidiana Ten 20 Apr '07

20 Apr '07

Hi there, I have some questions to ask about voicemail. I'm using SEMS voicemail to store voice message with SER v.0.9.6 and SEMS 0.10.0 I have my server running voicemail successfully. But I'm still confuse where voicemail store the message and in what form (text or wav file) I trace my voicemail storage and found out that it store the message in /var/spool/mail/ in text form. Is it true? So how to hear the voice message while it is in text file? And the other problem is, my voicemail server can only store message for root@localhost only. The script is like below: if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } } How to change this script so it can forward to user email address in SER database? Thanx before Regards Meidiana --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

3 2

[Serusers] sl_reply replies to source port, not contact port
by Mahatma Sarasvati 20 Apr '07

20 Apr '07

Hello, I'm using SER from cvs checked out today: cvs co -r rel_2_0_0 sip_routerMy Cisco 7940G phone is trying to register and is failing. I think because SER is replying to a the port (50650) that the REGISTER request came from, but the phone is not listening on port 50650. The phone never adds Digest Auth stuff into the REGISTER message, so I think its not getting the 401 Unauthorized message. The REGISTER message is explicit that the Contact should be at port 5060. I have a production installation of SER 0.8.14 that is replying to port 5060 on the phones with the 401 Unauthorized, so I know the phones work in that scenario. I'm using the default ser.cfg version 1.4 as posted on iptel.org in the Optomizing RTP Proxy page. See below. What to do? Thanks, Mahatma U 10.10.100.103:50650 -> 10.10.10.41:5060 REGISTER sip:velocity11.com SIP/2.0. Via: SIP/2.0/UDP 10.10.100.103:5060;branch=z9hG4bK5407bc35. From: <sip:gnoah@velocity11.com>;tag=0008e31b74c7001e396ea85f-3fa5879c. To: <sip:gnoah@velocity11.com>. Call-ID: 0008e31b-74c70006-2eef1c02-22c70ede(a)10.10.100.103. Max-Forwards: 70. Date: Thu, 19 Apr 2007 22:44:23 GMT. CSeq: 116 REGISTER. User-Agent: Cisco-CP7940G/8.0. Contact: <sip:gnoah@10.10.100.103:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0008e31b74c7>";+u.sip!model.ccm.cisco.com="8". Content-Length: 0. Expires: 3600. . U 10.10.10.41:5060 -> 10.10.100.103:50650 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.10.100.103:5060;branch=z9hG4bK5407bc35;rport=50650. From: <sip:gnoah@velocity11.com>;tag=0008e31b74c7001e396ea85f-3fa5879c. To: <sip:gnoah@velocity11.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad42. Call-ID: 0008e31b-74c70006-2eef1c02-22c70ede(a)10.10.100.103. CSeq: 116 REGISTER. WWW-Authenticate: Digest realm="velocity11.com", nonce="4627f1f527fd2a950a45686789ff5babccb80014", qop="auth". Server: Sip EXpress router (2.0.0-rc1 (i386/linux)). Content-Length: 0. Warning: 392 10.10.10.41:5060 "Noisy feedback tells: pid=7729 req_src_ip=10.10.100.103 req_src_port=50650 in_uri=sip:velocity11.com out_uri=sip:velocity11.com via_cnt==1". . __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] Openser accounting
by Lucas Diaz 20 Apr '07

20 Apr '07

Is it possible to save call length? Why can't I save BYE methods? Thanks.

2 1

[Users] radius accounting
by Terry Robert 19 Apr '07

19 Apr '07

Dear Openser, After some investigation, I discovered that the radius accounting behavior is different between openser-1.2.0-tls, and openser-1.1.1-tls. Specifically, when running the basic UAS/UAC scenario in SIPp, I observe the following: 1) Openser-1.1.1-tls - generates 2 accounting start packets, 1 accounting stop. User-Name, CalledStationId, CallingStationId attributes are present. 2) Openser-1.2.0-tls - generates 1 accounting start packet, 1 accounting stop. User-Name, CalledStationdId, CallingStationId attributes are missing. Can anyone: 1) confirm this behaviour? 2) tell me which one is correct? Thank you for your assistance in this matter.

3 2

[Users] symmetric firewall without nat
by Alexander Bergolth 19 Apr '07

19 Apr '07

Hi! In our organization there are many subnets with public-ip-adresses that are behind stateful-firewalls that prevent incoming connections from outside. If the clients are in two different protected subnets, they won't be able to communicate without an rtpproxy. Unfortunately since there is no NAT involved, the nat_uac_test() won't be useful. Is there any other way for the server to detect that an rtpproxy has to be used? Is there a way for the clients to detect it and report it to the server? Thanks in advance, --leo -- ----------------------------------------------------------------------- Alexander.Bergolth (at) wu-wien.ac.at Fax: +43-1-31336-906050 Zentrum fuer Informatikdienste - Wirtschaftsuniversitaet Wien - Austria

2 1

[Users] block calls
by Carlos Ribas 19 Apr '07

19 Apr '07

I need to block calls that they start with number 0. i did this, but it`s not OK: if (uri=~"^sip:[1-9]*@.*") { rewritehost("10.0.0.10"); log("access ok"); route(1); break; }; if (uri=~"^sip:0[1-9]*@.*") { log("access deny"); break; }; How can i do this?

2 2

[Users] TLS communication between OpenSER servers
by John Barry 19 Apr '07

19 Apr '07

Hi, I have configured two SIP domains using OpenSER v 1.1.1 with authentication (digest authentication) and TLS support. Each domain has clients (UAC) running minisip (3220) with TLS and certificates configured. When there is a call between a user from one domain to a user in the other domain, the SIP communications between minisip clients and their corresponding OpenSER proxies is done via TLS (port 5061/TCP), however, the SIP communication between the two OpenSER proxies is still done via UDP (port 5060/UDP). The TLS configuration in both servers is as follows: disable_tls = 0 listen = tls:192.168.1.10:5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/etc/openser/tls/sipdA/sipdA-cert.pem" tls_private_key = "/etc/openser/tls/sipdA/sipdA-privkey.pem" tls_ca_list = "/etc/openser/tls/sipdA/sipdA-calist.pem" Any ideas or suggestions regarding how to enable SIPS (TLS) between OpenSER SIP proxies? Thanks. JB74 _________________________________________________________________ Live Search, for accurate results! http://www.live.nl

2 1

[Serusers] Ser as IVR
by Arun Kumar 19 Apr '07

19 Apr '07

Hi, Is it possible to design an IVR using SER ? If yes please advice. thanks arun

2 1

[Users] Serweb + OpenSER 1.2.0 with postgres
by Joao Pereira 19 Apr '07

19 Apr '07

Hello is someone using serweb with OpenSER 1.2.0 and postgres? I want to know if it is possible. I tried the version in http://voip-info.org/wiki/view/OpenSER+1.1.x+and+SerWEB+0.9.4 but it isn't working and this error happens: Function aggregate_methods() doesn't exists. Try install Classkit extension. http://pecl.php.net/package/classkit I already tried classkit and the problem remains. Is someone using serweb + OpenSER 1.2.0 + postgres? Thanks regards Joao Pereira -- ______________________________________________ João Gomes Pereira FCCN Av. do Brasil, nº 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: joao.pereira(a)fccn.pt http://www.fccn.pt ______________________________________________ --- Aviso de Confidencialidade Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha recepcionado indevidamente esta mensagem, solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218 440 100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218 440 100 and delete it immediately. ---

2 2

Re: [Users] openser + nathelper + rtpproxy w/ ekiga
by Ben Racher 19 Apr '07

19 Apr '07

openser is not behind my NAT (see the dump, it has a public IP), wouldn't that defeat the purpose of nathelper? one of my clients is behind the NAT, the other isn't. some of the SDP flags appear to be correct, but I myself am still new enough to SIP that i can't tell what my SDP flags should be like coming out of the openser box. it appears to be correct, could you please take a look at the dump and tell me what specific transactions are causing it to fail? ben raviprakash sunkara wrote: > Hi > did U Check the sdp flags, nathelper modules in fixed_nated_sdp and > force_rtp_proxy flag,,,,,,, > > time interval to alive the natter.............. > > > is ur OpenSER is Behind the NAT, and Urs Users are also in Behind NAT > > > > On 4/19/07, *Ben Racher* <mack.stout(a)gmail.com > <mailto:mack.stout@gmail.com>> wrote: > > I'm trying to use openser's (1.1.1) default nathelper.cfg to > connect two > ekiga clients. One is behind a NAT. Both can contact each other, > but the > media streams (neither video nor audio) don't seem to be setup for > some > reason. rtpproxy registers that it should setup the connection, but it > doesn't seem to take off past that. > > here's an ngrep dump of the NAT'd client calling the other, and could > appreciate some wisdom on what I need to fix and what the problem is. > thanks. > > -------- > > U 2006/04/20 00:11:09.663917 209.91.15.175:5071 > <http://209.91.15.175:5071> -> 209.91.15.110:5060 > <http://209.91.15.110:5060> > INVITE sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110> SIP/2.0. > Date: Thu, 19 Apr 2007 04:44:30 GMT. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport. > > User-Agent: Ekiga/2.0.3. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110>>. > Contact: <sip:test_store@192.168.0.25:5071;transport=udp>. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Type: application/sdp. > Content-Length: 227. > Max-Forwards: 70. > . > v=0. > o=- 1176957870 1176957870 IN IP4 192.168.0.25 <http://192.168.0.25>. > s=Opal SIP Session. > c=IN IP4 192.168.0.25 <http://192.168.0.25>. > t=0 0. > m=audio 5006 RTP/AVP 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > m=video 5008 RTP/AVP 31. > a=rtpmap:31 H261/90000. > > > U 2006/04/20 00:11:09.673012 209.91.15.110:5060 > <http://209.91.15.110:5060> -> 209.91.15.175:5071 > <http://209.91.15.175:5071> > SIP/2.0 100 trying -- your call is important to us. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071;received=209.91.15.175 > <http://209.91.15.175>. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: < sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110>>. > Server: OpenSer (1.1.1-notls (i386/linux)). > Content-Length: 0. > Warning: 392 209.91.15.110:5060 <http://209.91.15.110:5060> "Noisy > feedback tells: pid=4047 > req_src_ip= 209.91.15.175 <http://209.91.15.175> req_src_port=5071 > in_uri=sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110> > out_uri=sip:test@209.91.15.19:5074;transport=udp via_cnt==1". > . > > > U 2006/04/20 00:11:09.673822 209.91.15.110:5060 > <http://209.91.15.110:5060> -> 209.91.15.19:5074 > <http://209.91.15.19:5074> > INVITE sip:test@209.91.15.19:5074;transport=udp SIP/2.0. > Record-Route: > <sip:209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > Date: Thu, 19 Apr 2007 04:44:30 GMT. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP 209.91.15.110 > <http://209.91.15.110>;branch=z9hG4bK393e.770a5756.0. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;received=209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > > User-Agent: Ekiga/2.0.3. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110>>. > Contact: <sip:test_store@209.91.15.175:5071;transport=udp>. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Type: application/sdp. > Content-Length: 270. > Max-Forwards: 69. > P-hint: usrloc applied. > . > v=0. > o=- 1176957870 1176957870 IN IP4 192.168.0.25 <http://192.168.0.25>. > s=Opal SIP Session. > c=IN IP4 209.91.15.110 <http://209.91.15.110>. > t=0 0. > m=audio 35000 RTP/AVP 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > m=video 35002 RTP/AVP 31. > a=rtpmap:31 H261/90000. > a=direction:active. > a=nortpproxy:yes. > > > U 2006/04/20 00:11:09.678496 209.91.15.19:5074 > <http://209.91.15.19:5074> -> 209.91.15.110:5060 > <http://209.91.15.110:5060> > SIP/2.0 100 Trying. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP 209.91.15.110 > <http://209.91.15.110>;branch=z9hG4bK393e.770a5756.0. > Via: SIP/2.0/UDP > 192.168.0.25:5071 <http://192.168.0.25:5071>;received= > 209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110>>. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Length: 0. > Record-Route: > <sip:209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > . > > > U 2006/04/20 00:11:09.801370 209.91.15.19:5075 > <http://209.91.15.19:5075> -> 209.91.15.110:5060 > <http://209.91.15.110:5060> > SIP/2.0 100 Trying. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP 209.91.15.110 > <http://209.91.15.110>;branch=z9hG4bK393e.770a5756.0. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;received=209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Length: 0. > Record-Route: > <sip:209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > . > > > U 2006/04/20 00:11:09.824859 209.91.15.19:5075 > <http://209.91.15.19:5075> -> 209.91.15.110:5060 > <http://209.91.15.110:5060> > SIP/2.0 502 Bad Gateway. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP 209.91.15.110 > <http://209.91.15.110>;branch=z9hG4bK393e.770a5756.0 . > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;received=209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Length: 0. > Record-Route: > <sip:209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > . > > > U 2006/04/20 00:11:09.827826 209.91.15.110:5060 > <http://209.91.15.110:5060> -> 209.91.15.19:5074 > <http://209.91.15.19:5074> > ACK sip:test@209.91.15.19:5074;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 209.91.15.110 > <http://209.91.15.110>;branch=z9hG4bK393e.770a5756.0. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > CSeq: 1 ACK. > User-Agent: OpenSer (1.1.1-notls (i386/linux)). > Content-Length: 0. > . > > > U 2006/04/20 00:11:09.828858 209.91.15.110:5060 > <http://209.91.15.110:5060> -> 209.91.15.175:5071 > <http://209.91.15.175:5071> > SIP/2.0 502 Bad Gateway. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;received=209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > From: "Test Store" > < sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Length: 0. > Record-Route: > <sip:209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > . > > > U 2006/04/20 00:11:09.847970 209.91.15.175:5071 > <http://209.91.15.175:5071> -> 209.91.15.110:5060 > <http://209.91.15.110:5060> > ACK sip:test@209.91.15.110 <mailto:sip:test@209.91.15.110> SIP/2.0. > CSeq: 1 ACK. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Length: 0. > Max-Forwards: 70. > . > > > U 2006/04/20 00:11:10.209491 209.91.15.19:5075 > <http://209.91.15.19:5075> -> 209.91.15.110:5060 > <http://209.91.15.110:5060> > SIP/2.0 200 OK. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP 209.91.15.110 > <http://209.91.15.110>;branch=z9hG4bK393e.770a5756.0. > Via: SIP/2.0/UDP > 192.168.0.25:5071 > <http://192.168.0.25:5071>;received=209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: < sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > Contact: <sip:test@209.91.15.19:5075;transport=udp>. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Type: application/sdp. > Content-Length: 153. > Record-Route: > <sip:209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > . > v=0. > o=- 1176929527 1176929527 IN IP4 209.91.15.19 <http://209.91.15.19>. > s=Opal SIP Session. > c=IN IP4 209.91.15.19 <http://209.91.15.19>. > t=0 0. > m=video 5012 RTP/AVP 31. > a=rtpmap:31 H261/90000. > > > U 2006/04/20 00:11:10.210568 209.91.15.110:5060 > <http://209.91.15.110:5060> -> 209.91.15.175:5071 > <http://209.91.15.175:5071> > SIP/2.0 200 OK. > CSeq: 1 INVITE. > Via: SIP/2.0/UDP > 192.168.0.25:5071 <http://192.168.0.25:5071>;received= > 209.91.15.175 > <http://209.91.15.175>;branch=z9hG4bK9280b754-9eec-db11-9dcc-00187185e000;rport=5071. > From: "Test Store" > <sip:test_store@209.91.15.110 > <mailto:sip:test_store@209.91.15.110>>;tag=a84faa54-9eec-db11-9dcc-00187185e000. > > Call-ID: 3a45aa54-9eec-db11-9dcc-00187185e000(a)192.168.0.25 > <mailto:3a45aa54-9eec-db11-9dcc-00187185e000@192.168.0.25>. > To: <sip:test@209.91.15.110 > <mailto:sip:test@209.91.15.110>>;tag=0a37f656-5cec-db11-9452-001372710463. > > Contact: <sip:test@209.91.15.19:5075;transport=udp>. > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. > Content-Type: application/sdp. > Content-Length: 173. > Record-Route: > <sip: 209.91.15.110 > <http://209.91.15.110>;lr=on;ftag=a84faa54-9eec-db11-9dcc-00187185e000>. > . > v=0. > o=- 1176929527 1176929527 IN IP4 209.91.15.19 <http://209.91.15.19>. > s=Opal SIP Session. > c=IN IP4 209.91.15.110 <http://209.91.15.110>. > t=0 0. > m=video 35004 RTP/AVP 31. > a=rtpmap:31 H261/90000. > a=nortpproxy:yes. > > > > _______________________________________________ > Users mailing list > Users(a)openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > www.hyperion-tech.com <http://www.hyperion-tech.com> > Client and Parent company :- www.august-networks.com > <http://www.august-networks.com>

1 0

[Users] presence: dealing with publish
by Cesc 19 Apr '07

19 Apr '07

Hi, I am using openser1.2 If the presence module receives a PUBLISH: - without E-Tag, it gets stored and if needed, a NOTIFY is generated ... works for me. - with an E-tag, for which this openser has not seen the first PUB (the one without E-tag) ... it gets discarded/ignored? How is the e-tag used in processing publishes and then generating notifies? I ask this because in my (peculiar) setup with two openser's sending publish to each other, depending on the order I start the elements (2 phones, 2 opensers), I don't get always a working behavior: - if the opensers are running before the phones start ... no problem ... I guess as they see the PUB without e-tag, it all goes fine - if the boot order is : openser1 ... then phone 1 ... sometime after (enough to miss the first PUB without e-tag) ... openser2 ... then phone2 ... Phone 2 does not receive NOTIFYs of status changes from phone1, eventhough the PUBlish messages do get to openser2 ... but somehow, they get "discarded" ... Regards, Cesc

2 1

[Users] corrupted From with uac_replace_from in case of re-invite
by tele 19 Apr '07

19 Apr '07

Hi all, I'm occured in this problem already seen here: http://www.openser.org/pipermail/users/2006-March/003540.html My problem occured between two different UA and only in case of a Re-INVITE for a T.38 fax communication. Im' using openser 1.2.0. I'm doing some translation with From for a correct CLI screening. I'm also checked that the RR parameters are correctly mirrored. find attached here the debug.log and a network trace. thank you :tele

2 8

[Users] Using OpenSER as SIP Proxy over NAT
by Saar Carmi 19 Apr '07

19 Apr '07

Hi I need to use the SER proxy over NAT. For that, I only need that all SIP communcation between SER and other Sip Server will always work on a single port - altough the SIP client that connects SER use different port for REGISTER and INVITE Is it possible? Thank you Saar Carmi -- Saar Carmi Mobile: 054-7782417 Email: saarcarmi(a)gmail.com

2 1

[Serusers] module documentation - new search page
by Marcin Pazdro 19 Apr '07

19 Apr '07

Hi all, I would like to encourage you to test and use new module search page. It was placed here http://www.iptel.org/ser/doc/search Additionally some links to this page were added by greger in appropriate places across the iptel.org portal. Note that search-page is constructed using AJAX "technology" and used frameworks limit the amount of browsers that may correctly display this page. Supported browsers are: - IE >= 6; - Opera >= 9; - FF >= 1.6; Should you experience any problems or if you have any comments or suggestions, please let me know. If you are unable to use it (because of browser) and you would like to have a search page that support even lynx and have similar functionalities like described in this functional specification: http://www.iptel.org/ser_documentation_dynamic_search_page_functional_speci…, write to me too. If there will be huge demand for less interactive but more universal and better supported version of that search page I will consider creating it. Best regards, Marcin :)

3 2

[Users] IBM machines Supports the OpenSER as production
by raviprakash sunkara 19 Apr '07

19 Apr '07

Hello Users, I want to know , is OpenSER supports Highly expensive IBM machines for the VoIP services. the Machines like IBM System p™ servers IBM IntelliStation(r) POWER™ IBM BladeCenter(r) IBM System i5™, eServer™ i5 and eServer™ iSeries™ systems These are Unix I need to install Fedora Core 6 or 5 or SUSE linux into them, Please help me ........................ -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

2 3

[Users] NEW FEATURE: DNS-based failover
by Bogdan-Andrei Iancu 19 Apr '07

19 Apr '07

Hi everybody, OpenSER 1.2.0 has now the capability to do DNS based failover, according to RFC3263 (http://www.ietf.org/rfc/rfc3263.txt). The SIP resolver was enhanced to to be able to save and resume later the DNS queries in order to get all possible IP destinations. The resolving process is step-by-step done (get next IP only on demand) to minimize the total number of DNS queries. So having this support does not imply, in normal processing, more load on the DNS server. Additional queries are done only when needed (after failure detected). The scanning for new IP destinations is done by the SIP resolver on all DNS levels: NAPTR, SRV, A. Both core and TM are using this new feature. In core, the stateless forwarding can do only DNS-based failover at transport level (if no egress interface found or send operation failed due whatever reason). In TM, the DNS-based failover is extended to transaction level. If the transaction completes with 503 or 408 with no reply, automatically, a new branch will be fork if any destination IP can be found by the DNS resolver. Read more here http://www.openser.org/docs/modules/1.2.x/tm.html#AEN103 For controlling this feature use: - newly added core parameter "disable_dns_failover" - use to generally disable the DNS-based failover. By default is false. - in TM, t_relay() take a new flag for turning off the DNS-based failover. This setting is per transaction. By default, the failover is done. Any feedback is appreciated. regards, Bogdan

4 9

[Users] avp_db_query doesn't work with DBTEXT???
by Daryl Sanders 19 Apr '07

19 Apr '07

I cannot seem to get avp_db_query to work with dbtext. My query looks like this.. avp_db_query("SELECT uuid FROM subscriber WHERE username = '$rU'", "$avp(s:callee_uuid)"); I receive the following error: avpops:db_query_avp: error - cannot do the query Are there any other options for doing a query on a dbtext table from within the config? Thanks, Daryl

2 1

[Users] nathelper openser 1.2, change all c= in SDP
by laurent schweizer 19 Apr '07

19 Apr '07

Hello, I use opsnser 1.2 with nathelper and I see that not all c= in the SDP are changed. with some user agent it's ok but for some else, this is a problem. any solutions ? flag ? Laurent exemple: SDP in the 200 ok modified with nathelper (openser -> UA ) v=0.o=CiscoSystemsSIP-GW-UserAgent 5635 2443 IN IP4 xxx.xxx.127.1.s=SIP Call.c=IN IP4 xxx.xxx.127.1.t=0 0.m=audio 43594 RTP/AVP 8 106.c=IN IP4 yyy.yyy.14.234.a=rtpmap:8 PCMA/8000.a=rtpmap:106 telephone-event/8000.a=fmtp:106 0-15.a=ptime:20.a=nortpproxy:yes..

2 1

[Serusers] SER for Multimedia Gateway with PCI S/T Interface
by Sahria Hao 19 Apr '07

19 Apr '07

Hi all, I want to build gateway server using SER and PCI S/T interface (PRI or COT). Can SER use for gateway server like (Asterisk + Digium) ? Regards, Sahria

2 1

[Serusers] set timer to drop the call
by jorain 18 Apr '07

18 Apr '07

Hi all, Is there any function or method to set timer in ser.cfg? When the timer end, the call will be terminated. Thanks, jorain

2 1

[Users] Re: null radius_extra avp for acc is causing openser to crash
by Ovidiu Sas 18 Apr '07

18 Apr '07

Does anyone else experienced this? Regards, Ovidiu Sas On 4/11/07, Ovidiu Sas <sip.nslu(a)gmail.com> wrote: > Hi all, > > If an avp that is defined as an extra attribute for radius accounting > is null or not defined, it is causing openser to crash. The code > should check if the avp or PV is null and drop or send the attribute > based on value. > > > Regards, > Ovidiu Sas >

2 7

[Users] Reg:call forwarding problem
by venkid 18 Apr '07

18 Apr '07

Dear all, i am trying forwarding call with openser .i am using mediaproxy with openser. call is not getting forwarded uri.when i call UA1 - UA2 same call have to forwarded to UA3 but two INVITE messages are coming to UA2 . call is not going to UA3 .i am sending openser.cfg file also . please help me on this. at line 190 in openser.cfg am trying to forward using append_branch() . Thanks and Regards venkatesh.d debug=6 fork=yes log_stderror=yes listen=192.168.0.248 port=5060 children=4 #dns=no #rev_dns=no #debug=9 # debug level (cmd line: -ddddddddd #fork=yes #log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes #check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) #port=5060 #children=4 #fifo="/tmp/openser_fifo" fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openser@localhost/openser" 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.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" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/presence.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") 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/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #modparam("rr", "enable_full_lr", 1) modparam("presence", "db_url","mysql://openser:openser@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') modparam("presence", "totag_avpid", 987) modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "force_active", 1) #modparam("avpops", "avp_aliases", "uuid=I:50") route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("sip.pyrogroup.com:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="ACK") { route(1); exit; } else if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); ###Call Forwaring here append_branch("sip:shekhar@58.68.74.84"); if (!t_relay()) { # if (method="INVITE" || method=="ACK") { # end_media_session(); # }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("localhost","subscriber")) { www_challenge("localhost","0"); #consume_credentials(); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } 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(); }; }

1 0

[Users] Adding headers to the CANCEL messages generated by openser
by Ardjan Zwartjes 18 Apr '07

18 Apr '07

Hello openser users, I'm trying to do something with openser but I haven't been able to figure out how to do this. So I hope somebody on the list knows the answer: In my configuration script I use append_hf to add application data to the SIP messages, this works great until the moment where I do a parallel branch. The invites work just fine since I can do the append_hf in the branch_route. The problem arises when one of the branches is answered, at this point OpenSER will generate CANCEL messages for the other branches and I have no clue on how to add header fields with information about the original invite to those messages. So my question is: Does somebody know how you can add headers with data to the CANCEL messages generated by SER in a parallel branch? Thanks in advance. Kind regards, Ardjan Zwartjes.

2 4

[Users] Diferences between "fix_contact" and "fix_nated_contact"
by Edson 18 Apr '07

18 Apr '07

Hi, Despite the fact that this two functions are declared on NATHelper and Mediaproxy modules, from a practice point of view, they appear to do the same, so when to apply one and another? Aren't there functionalities overlaped? Edson. ------ "It wouldn't be sell if it were easy..." (myself)

2 1

[Users] prefix in dispatcher
by Jayesh Nambiar 18 Apr '07

18 Apr '07

Hi, I am using dispatcher module in my openser 1.2. I am using ds_select_domain function in the script. What I wanted to know is, is there a way that I can prefix some digits to the ruri before sending to the selected domain. My scenario is: I have got two Gateway IPs in the dispatcher module. Each Gateway needs a different prefix before the ruri to process the calls properly. I am using the round robin algo in dispatcher. Now the question is how will I know what IP is being selected and how will I attach appropriate prefix? Is there a way to do the same in the dispatcher module. My configuration snippet: route[4] { #--Dispatcher route-- ds_select_domain("1", "4"); $avp(i:25) = 3; $avp(i:30) = $rd; route(1); exit; } thanks in advance. w/regards, Jayesh Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/

2 1

[Users] avp_check() replacement? Openser 1.2?
by Brandon Armstead 18 Apr '07

18 Apr '07

Hello, I'm wondering if in OpenSER 1.2 if there is a replacement method to check if a value is inside an avp list, as follows: if($avp(some_avp_value) == "$avp(some_avp_list_value)/g"){ # do something } avp check equivlent: if(avp_check("$avp(some_avp_value)", "eq/$avp(some_avp_list_value)/g"){ # do something } Does anyone know if this is implemented or being planned for implementation, what are everyone else's thoughts on something like this? Thanks! -- Brandon Armstead

2 1

[Users] OpenSER status meeting 20-04-2007 13:00 GMT
by Henning Westerholt 18 Apr '07

18 Apr '07

Hello all, the first IRC OpenSER status meeting will take place this friday, the 20-04-2007 at 13:00 GMT (15:00 CEST) at the #openser channel on irc.freenode.net. The purpose of this meeting is to discuss the latest developments in and around the OpenSER project. Topics so far: 1. Experiences with OpenSER 1.2 2. OpenSER project server hosting 3. State of OpenSER Documentation 4. Next release For information about the IRC meetings read: http://openser.org/dokuwiki/doku.php/development:irc-meetings For details about this meeting read: http://openser.org/dokuwiki/doku.php/development:irc-meeting-agenda-20-04-2… Please join us! -- Henning Westerholt

2 1

[Users] extraneous 408's
by Tim 18 Apr '07

18 Apr '07

Hello, Since upgrading to 1.2 we have begun to see lots of extraneous 408 messages. The CSEQ indicates they are associated with INVITE transactions. Weird thing is they happen well after the INVITE transaction is complete. Any ideas on how I can figure out what is going on here? These are generated by openser not proxied from another source. Tim

2 2

Re: [Serusers] interesting 2 way SIP signalling problem
by Ladislav Andel 18 Apr '07

18 Apr '07

Hi, Looks like you are mixing together nathelper and mediaproxy module. You need to call force_rport() even for INVITEs. Look at Getting started ser configs ftp://siprouter.onsip.org/pub/gettingstarted/configs/ Lada İlker Aktuna (Koç.net) wrote: > Hi, > > Did you receive my reply yesterday ? > > Thanks, > ilker > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) > Sent: Monday, April 09, 2007 11:29 AM > To: Ladislav Andel > Cc: serusers(a)iptel.org > Subject: RE: [Serusers] interesting 2 way SIP signalling problem > > Hi, > > This is under my method=="REGISTER" section: > > if (nat_uac_test("1")) { > setflag(6); > fix_nated_contact(); > force_rport(); > }; > > > I also have this under method=="INVITE" : > > if (method=="INVITE" || method=="ACK") { > > if (client_nat_test("19")) { > fix_nated_contact(); > }; > > And this one under a onreply route : > > onreply_route[2] { > if (status=~"(183)|(2[0-9][0-9])") { > use_media_proxy(); > > if (nat_uac_test("19")) { > fix_nated_contact(); > }; > } > }; > > > Thanks for your reply... > > ilker > > -----Original Message----- > From: Ladislav Andel [mailto:ladaan@iptel.org] > Sent: Monday, April 09, 2007 11:31 AM > To: İlker Aktuna (Koç.net) > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] interesting 2 way SIP signalling problem > > Hi, > > do you call force_rport() function for NATed clients? > Can you paste in the NAT section of your ser.cfg here? > Paste in NAT detecting section too. > > Lada > > > İlker Aktuna (Koç.net) wrote: > >> Hi, >> >> I have a working configuration on my SER server. Most of the users can make calls (both way incoming and outgoing) with no problem. >> However, I have some clients that can't make calls. >> When I analyze these by ngrep and/or tcpdump, I see that the SER server does not reply the packet on the correct port. >> >> For example, I have a testing client which is behind NAT. >> The client sends INVITE message from UDP port 3023 to port 5061 of my SER. >> (cseq number is 1) >> SER exchanges the message with the uplink sip gateway and when it >> receives a "OK" response it forwards this message to the client's NAT >> IP address BUT on a different port (14640) with correct cseq number >> (1) >> >> Of course, this new port is not open on the NAT router and it does not reach the client. >> >> I see that the first INVITE message has Contact field with internal IP address of the client and the port number as 14640. >> So the SER server sends the returning message to the external IP address with internal port number. >> >> How can I fix this issue ? >> >> I can send the configuration of my SER server if that would help to find the problem. >> > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _____________________________________________________________________________________________________________________________________________ > Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. > This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. > _____________________________________________________________________________________________________________________________________________ >

3 4

[Serusers] how-to "loop" in the ser.cfg file
by Steve Blair 17 Apr '07

17 Apr '07

Hello: I need to perform a check several times within a failure_route and I'm not sure how to accomplish this task. The way I implemented sequentially hunting I perform the relay to the first number in the hunt list within route block 0. If this number is unavailable I "fail" to a failure block then interate through the remaining numbers in the huntlist in that failure route but this presents a problem. All of the "feature" and "call permission" checks which I normally perform on an extension are only performed in route block 0. This means I am not checking the feature and call permission flags for the 2nd through Nth number in the huntlist once I enter the failure route. I created a route block to perform the necessary checks but the problem is I do not see how to return to the failure route if the feature check for a given number fails. For example. If the t_relay to extension 1234 fails I will enter failure_route #1. In failure_route #1 I load the next extension, 2345, but if the call permission check for 2345 fails how do I skip this number in the huntlist, load the next number in the list and repeat the failure_route #1 processing? Thanks,Steve

3 4

[Users] SEMS: ivr: mp3 module
by Helmut Kuper 17 Apr '07

17 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have sems with voicemail module and mp3 support running. I modified it to save recorded mp3 files on harddisc. I modified ivr's mailbox_query module to load that prevoiusly on harddisc saved mp3 files. Unfortunately the male mp3 library isn#t able to decode/read/play that mp3 files. Is there any change that there will be mp3 decoding support in mp3 module? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGJOTm4tZeNddg3dwRAhxdAJ9UjRKha1hqV52MN70uwagyLMawVACfQwz8 FnrVEKj0ln45hZmlA9S2xYk= =2q7+ -----END PGP SIGNATURE-----

2 1

[Serusers] question re SER capacity and redirecting using rURI contents.
by Tom Meadows 17 Apr '07

17 Apr '07

hi- I am looking into using SER as part of our test infrastructure for a SIP-based messaging system and have 2 questions: one about SER capacity, and one about the best way to configure it. Firstly though, the setup: We use a script-driven inhouse SIP/RTP test tool in association with our messaging system which can either send or accept SIP invitations and process the associated RTP streams (eg act as UAS or UAC). This tool has a capacity of ~125 ports. We need the messaging system to create around 800 concurrent SIP/RTP sessions (dial out), and our system allows only one outbound sip gateway. I therefore need to configure a sip redirector which can take incoming SIP invites from the messaging system and redirect them to one of 7 test tool clients (125x7 > 800). First question, re capacity - the 800 concurrent sessions will be staggered and spread over 2 minutes. This works out at about 7 SIP invites per second. Should SER be able to handle this (proposed SER box =2.8Ghz, 512mB RAM). Secondly, the Request-URIs of the SIP invites sent by the messaging system are structured, and it is easy to tell from the Request-URI which client machine we want this invite to be redirected to. Eg: 1000000000-1000000999(a)dom.com -> should go to client 1 1000001000-1000001999(a)dom.com -> should go to client 2 1000002000-1000002999(a)dom.com -> should go to client 3 1000003000-1000003999(a)dom.com -> should go to client 4 etc Rather than registering all these accounts on SER, is there an alternative way I can get SER to redirect various SIP invites using wildcards to check the structure of their r-URI? eg 1000000...(a)dom.com redirects to A 1000001...(a)dom.com redirects to B 1000001...(a)dom.com redirects to C etc. Thanks for advice. Tom _________________________________________________________________ Txt a lot? Get Messenger FREE on your mobile. https://livemessenger.mobile.uk.msn.com/

2 1

[Serusers] IVR module can't be built
by Meidiana Ten 17 Apr '07

17 Apr '07

Hi there, Thanx before for my past questions. Now, I want to add mailbox application on my server but I cannot find IVR plugin Then, I read some tutorial that I must make the plugin itself I've followed the step but it failed. What should I do then? Please advise Thanx Regards, Meidiana --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

1 0

[Users] LCR failure_route appending characters to invite
by Lyle Thaut 17 Apr '07

17 Apr '07

Hello All, I'm having problems when calling next_gw from a failure_route. It appears that extra characters are getting appended after the transport=udp portion of the INVITE. This first INVITE that goes out on the wire looks normal. I'm intentionally causing the 1st gateway to be offline (time out) so it failure routes to the next gw: SIP/SD Request: INVITE sip:1310xxxyyyy@192.168.18.13:5060;transport=udp The next INVITE to the next gw (called from failure_route) fails due to extra characters. Note the \000dp after the transport=udp: SIP/SD Request: INVITE sip:1310xxxyyyy@192.168.142.40:5060;transport=udp\000dp And the same thing happens on the next failure route with the \0009b SIP/SD Request: INVITE sip:1310xxxyyyy@192.168.101.202:5060;transport=udp\0009b Looks like I've got extras characters on each of the bad INVITES. \0009b = .9b & \000dp = .dp A sample config that reproduces the problem looks like this: # ------------------------------------------------------------- route{ ...snip... route(1); } # ------------------------------------------------------------- route[1] { if(!load_gws()){ sl_send_reply("500", "Internal server error\n"); xlog("LCR Failure: Unable to load gateways\n"); exit; }; if(!next_gw()){ sl_send_reply("503", "Service not available"); xlog("LCR Failure: No gateways found\n"); exit; }; t_on_failure("1"); # Process Failures if (!t_relay()) { sl_reply_error(); }; exit; } # ------------------------------------------------------------- failure_route[1]{ if(!next_gw()){ t_reply("503", "Service not available"); xlog("$CwrLCR Failure: No gateways found$Cxx\n"); exit; }; t_relay(); exit; } # ------------------------------------------------------------- If I replace the next_gw call in the failure_route with a rewrite_uri(), append_branch() and a t_relay() then it works as desired so I believe that the problem is LCR related. Debug shows the URI's seem to be correct in the gw_uri_avp: 2(12902) load_gws(): DEBUG: Added gw_uri_avp <sip:1|0@192.168.101.202:5060;transport=udp> 2(12902) load_gws(): DEBUG: Added gw_uri_avp <sip:1|0@192.168.142.40:5060;transport=udp> 2(12902) load_gws(): DEBUG: Added gw_uri_avp <sip:|0@192.168.18.13:5060;transport=udp> System: Solaris10/Sparc64 OpenSER: 1.2.0 with TLS Any suggestions or thoughts? Thanks, Lyle

3 3

[Users] ds_select_dst("1","0"): hash on callid
by Mark Price 17 Apr '07

17 Apr '07

Hi, I'm having some interesting results with using the ds_select_dst function to load balance asterisk boxes. I.e. packets with method==INVITE are routed with ds_select_dst. when using two destinations in the dispatcher.txt file, the calls get disproportionately sent to one destination, unless i'm using the callid hash. However callid's property of changing with every invite (reinvite, etc) causes me to question it's suitableness. For example: A calls B, puts B on hold, takes B off hold. Error prone or not? A calls B, puts B on hold, calls C, conferences. Ad nauseum. Am I missing something? Is there a better approach? Should it Just Work, even hashing on callid? Mark Price

2 1

[Users] Openser 1.1.0 Debian and OSF module
by Howard Tang 17 Apr '07

17 Apr '07

Hi All, I found that the Openser 1.1.0 Debian package doesn't install the OSF module. I would like to make use of the OSF module now, what will the best way to get the OSF module? can i just simple do apt-get install openser again to upgarde to 1.1.1 ? will that come with OSF module? Since the system is currently production, i want to avoid any downtime or server reboot. Regards, Howard

2 3

[Users] MI_XMLRPC Compilation problem on OpenSUSE 10.2 - 64 bits
by Edson 16 Apr '07

16 Apr '07

Hi, Folks... When I try to compile OpenSER 1.2.0 with MI_XMLRPC support, I get this error: mi_xmlrpc.c:44:26: error: xmlrpc_abyss.h: No such file or directory mi_xmlrpc.c: In function âmod_child_initâ: mi_xmlrpc.c:162: warning: implicit declaration of function âxmlrpc_server_abyss_init_registryâ mi_xmlrpc.c:172: error: âxmlrpc_server_abyss_rpc2_handlerâ undeclared (first use in this function) mi_xmlrpc.c:172: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:172: error: for each function it appears in.) mi_xmlrpc.c:178: error: âxmlrpc_server_abyss_default_handlerâ undeclared (first use in this function) make[1]: *** [mi_xmlrpc.o] Error 1 I know that its because of the lack of the xmlrpc_abyss.h file, but my problem is where to find this file? In which package did it reside? As sad on the subject Im using OpenSUSE 10.2 - 64 bits on a Xeon machine. Edson. ------ "It wouldn't be sell if it were easy..." (myself)

2 2

[Users] lcr for failover broken after upgrade to 1.2
by Brandon Price 16 Apr '07

16 Apr '07

I was using lcr for failover with a route that looks like.... ------------------------------------------------------------------------ ------------ { if(!t_check_status("200")){ xlog("got to failure route 2 lcr"); if (!next_gw()) { xlog ("503, service not available - no more gateways"); t_reply("503", "Service not available - no more gateways"); exit; } else { t_on_failure("2"); xlog("lcr forwarding invite from $fu to -> $ru"); t_relay(); return; }; }; } ------------------------------------------------------------------------ ------------ The t_relay after the line that says xlog("lcr forwarding invite from $fu to -> $ru"); no longer works. I can see the correct $fu and $ru values in the xlog output but the t_relay is never done. For example.. my.sip.proxy.ip -> my.pstn.gw.ip SIP/SDP Request: INVITE sip: +13143212222@my.pstn.gw.ip:5060;transport=udp, with session description my.pstn.gw.ip -> my.sip.proxy.ip SIP Status: 100 Trying my.pstn.gw.ip -> my.sip.proxy.ip SIP Status: 604 Invalid Number Format my.sip.proxy.ip -> my.pstn.gw.ip SIP Request: ACK sip: +13143212222@my.pstn.gw.ip:5060;transport=udp just ends there where as witn 1.1 it would then forward the invite to the next gw in the lcr table. Any help would be greatly appreciated.

1 0

[Users] MI_XMLRPC Compilation problem on OpenSUSE 10.2 -64
by Edson 16 Apr '07

16 Apr '07

Hi, Folks... When I try to compile OpenSER 1.2.0 with MI_XMLRPC support, I get this error: mi_xmlrpc.c:44:26: error: xmlrpc_abyss.h: No such file or directory mi_xmlrpc.c: In function âmod_child_initâ: mi_xmlrpc.c:162: warning: implicit declaration of function âxmlrpc_server_abyss_init_registryâ mi_xmlrpc.c:172: error: âxmlrpc_server_abyss_rpc2_handlerâ undeclared (first use in this function) mi_xmlrpc.c:172: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:172: error: for each function it appears in.) mi_xmlrpc.c:178: error: âxmlrpc_server_abyss_default_handlerâ undeclared (first use in this function) make[1]: *** [mi_xmlrpc.o] Error 1 I know that its because of the lack of the xmlrpc_abyss.h file, but my problem is where to find this file? In which package did it reside? As sad on the subject Im using OpenSUSE 10.2 - 64 bits on a Xeon machine Edson Gellert Schubert Phone :(0xx47) 3422-6556 Celular:(0xx47) 9107-6556 MSN :egschubert@hotmail.com Brasil - (GMT-3)

1 0

[Users] rtpproxy - ICMP unreacheable host
by Amanda Mattiuz 16 Apr '07

16 Apr '07

Hello! I'm using rtpproxy to handle NAT, but right now I have problems of unreacheable host... I believe client's firewalls are blocking UDP traffic coming from the proxy ... I tried to change port range and nothing seem to make it work... I was wondering if any of you kows something that may fix it.... Thanks -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

3 10

[Users] usrloc, presence etc. cleanup rate: OpenSER mysql backend
by Bernard Chan 16 Apr '07

16 Apr '07

Hello, I would like to know how frequently OpenSER are cleaning up stale records with the mysql backend of various modules? For instance, active_watchers, location, presentity, watchers etc. mysql tables, or are stale records automatically removed at all? As I found a row in the location table with an "expires" date more than a month ago, but was still not cleaned up today. Or OpenSER takes a different strategy? This is because we expect to deploy OpenSER on a fairly large network and we are estimating database size and related overhead, and we are a bit worry that accumulation of these stale records may increase our processing costs. Thank you in advance for any insights. Regards, Bernard Chan.

3 6

Re: [Users] caller and callee behind diff NAT - ACK of 200OK is not reaching to callee
by shankar 16 Apr '07

16 Apr '07

I am very sorry Bogdan due to comany policy I can not attach any file. Regards, Shankar ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: shankar <shankar_bandal(a)yahoo.com> Cc: users(a)openser.org Sent: Monday, 16 April, 2007 1:29:56 PM Subject: Re: [Users] caller and callee behind diff NAT - ACK of 200OK is not reaching to callee Hi there, unfortunately your msg flow is not comprehensive due text misalignment. Make it a simple text file and attach it to the email . regards, bogdan shankar wrote: > Hi ALL, > > Currently I am useing ser 0.9.6 version, I am switching to openser but before that I want to find out whether my scenario is valid. If it is valid then how can I solve this problem using openser. > > Belo is message sequence for my scenario. > > user1 NAT router 1 SER Proxy SIP Server NAT router 2 User2 > (192.x.x.3) (107.x.x.78) (107.x.x.201) (107.x.x.150) (107.x.y.249) (192.x.x3) > | REGISTER(user1) |REGISTER(user1) |REGISTER(user1)| | | > |-------------------> |---------------> |--------------> | | | > | | |REGISTER(user2)| | | > | | | ------------> | | | > | | | REGISTER(user2) | REGISTER(user2)| > | | | <--------------------------------------------------------|<------------------------ | > | INVITE | INVITE | INVITE | INVITE | INVITE | > |----------------------------->|-------------------------->| ----------------------->|------------------------------>|------------------------->| > | | | | | | > | 100 trying | 100 trying | | | | > | <-------------------------- |<------------------------- | | | | > | 180 Ringing | 180 Ringing | 180 Ringing | 180 Ringing | 180 Ringing | > | <--------------------------- | <----------------------- |<---------------------- | <---------------------------- | <---------------------- | > | | | | | | > | 200 OK | 200 OK | 200 OK | 200 OK | 200 OK | > | <--------------------------- | <---------------------- | <---------------------- | <---------------------------- | <---------------------- | > | | | | | | > | ACK | ACK | ACK | | | > | -----------------> | -----------------------> | ----------------------> | ?????? | | > Set up: > user 1: > Internal IP: 192.x.x.3 > external ip: 107.x.x.78 > user 2: > Internal IP: 192.x.x.3 > external ip: 107.x.y.249 > > Nat routers are FC5 linux PCs on which I have enabled SNAT and posrt forwarding. > > In above case user2 is not getting ACK. fix_nated_contact() API of SER is modifying contact > header to the destination ip of receiving message. e.g. SER proxy receives REGISTER meassage > from user 1 from NAT router 1 then it changing contact ip to the dst ip of REGISTER message i.e. > Nated ip of user1. Because of this RFEGISTER and INVITE goes fine. > But in case of 200 OK of INVITE comming from user 2 dst ip is IP of SIP server, so SER adds server ip in the contact header of 200Ok because of this user1 is sending ACK to SERVER but not to user2. > If I ignore the ACK (just for testing) and start media using RTPPROXY then media packets are > going to SERVER but not to user2 because force_rtp_proxy follows the same principle like > fix_nated_contact(). > > Please provide me some pointers. I am not able to unserstand how use1 will know that 200OK is came from user 2. In 200OK message there no reference of the nated IP of user2 > > below is my SER config file. > # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ > debug=6 > fork=no > log_stderror=yes > listen=107.108.70.201 # INSERT YOUR IP ADDRESS HERE > port=5060 > children=4 > dns=no > rev_dns=no > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser:heslo@localhost/ser" > #unix_sock="/tmp/ser_sock" > 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/auth.so" > #loadmodule "/usr/local/lib/ser/modules/auth_db.so" > loadmodule "/usr/local/lib/ser/modules/uri.so" > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > #loadmodule "/usr/local/lib/ser/modules/domain.so" > #loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > #modparam("auth_db", "calculate_ha1", 1) > #modparam("auth_db", "password_column", "password") > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > modparam("usrloc", "db_mode", 2) > modparam("registrar", "nat_flag", 6) > modparam("rr", "enable_full_lr", 1) > route { > # ----------------------------------------------------------------- > # Sanity Check Section > # ----------------------------------------------------------------- > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483", "Too Many Hops"); > break; > }; > if (msg:len > max_len) { > sl_send_reply("513", "Message Overflow"); > break; > }; > # !! Nathelper > # Special handling for NATed clients; first, NAT test is > # executed: it looks for via!=received and RFC1918 addresses > # in Contact (may fail if line-folding is used); also, > # the received test should, if completed, should check all > # vias for rpesence of received > if (nat_uac_test("3")) { > # Allow RR-ed requests, as these may indicate that > # a NAT-enabled proxy takes care of it; unless it is > # a REGISTER > if (method == "REGISTER" || ! search("^Record-Route:")) { > log("LOG: Someone trying to register from private IP, rewriting\n"); > # This will work only for user agents that support symmetric > # communication. We tested quite many of them and majority is > # smart enough to be symmetric. In some phones it takes a configuration > # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is > # called "symmetric media" and "symmetric signalling". > fix_nated_contact(); # Rewrite contact with source IP of signalling > # if (method == "INVITE") { > # fix_nated_sdp("1"); # Add direction=active to SDP > # }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!method=="REGISTER") record_route(); > # ----------------------------------------------------------------- > # Call Tear Down Section > # ----------------------------------------------------------------- > if (method=="BYE" || method=="CANCEL") { > unforce_rtp_proxy(); > }; > # ----------------------------------------------------------------- > # Loose Route Section > # ----------------------------------------------------------------- > if (loose_route()) { > if ((method=="INVITE" || method=="REFER") && !has_totag()) { > sl_send_reply("403", "Forbidden"); > break; > }; > if (method=="INVITE") { > # if (!proxy_authorize("","subscriber")) { > # proxy_challenge("","0"); > # break; > # } else if (!check_from()) { > # sl_send_reply("403", "Use From=ID"); > # break; > # }; > # consume_credentials(); > > if (nat_uac_test("19")) { > setflag(6); > force_rport(); > fix_nated_contact(); > }; > force_rtp_proxy("l"); > }; > route(1); > break; > }; > # ----------------------------------------------------------------- > # Call Type Processing Section > # ----------------------------------------------------------------- > if (uri!=myself) { > route(4); > route(1); > break; > }; > if (method=="ACK") { > route(1); > break; > } else if (method=="CANCEL") { > route(1); > break; > } else if (method=="INVITE") { > route(3); > break; > } else if (method=="REGISTER") { > route(2); > break; > }; > lookup("aliases"); > if (uri!=myself) { > append_hf("P-hint: outbound alias\r\n"); > route(4); > route(1); > break; > }; > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > break; > }; > route(1); > } > route[1] { > # ----------------------------------------------------------------- > # Default Message Handler > # ----------------------------------------------------------------- > t_on_reply("1"); > if (!t_relay()) { > if (method=="INVITE" && isflagset(6)) { > unforce_rtp_proxy(); > }; > sl_reply_error(); > }; > } > route[2] { > # ----------------------------------------------------------------- > # REGISTER Message Handler > # ---------------------------------------------------------------- > if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > sl_send_reply("100", "Trying"); > # if (!www_authorize("","subscriber")) { > # www_challenge("","0"); > # break; > # }; > # if (!check_to()) { > # sl_send_reply("401", "Unauthorized"); > # break; > # }; > # consume_credentials(); > if (!save("location")) { > sl_reply_error(); > }; > } > > > Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/

2 1

[Users] caller and callee behind diff NAT - ACK of 200OK is not reaching to callee
by shankar 16 Apr '07

16 Apr '07

Hi ALL, Currently I am useing ser 0.9.6 version, I am switching to openser but before that I want to find out whether my scenario is valid. If it is valid then how can I solve this problem using openser. Belo is message sequence for my scenario. user1 NAT router 1 SER Proxy SIP Server NAT router 2 User2 (192.x.x.3) (107.x.x.78) (107.x.x.201) (107.x.x.150) (107.x.y.249) (192.x.x3) | REGISTER(user1) |REGISTER(user1) |REGISTER(user1)| | | |-------------------> |---------------> |--------------> | | | | | |REGISTER(user2)| | | | | | ------------> | | | | | | REGISTER(user2) | REGISTER(user2)| | | | <--------------------------------------------------------|<------------------------ | | INVITE | INVITE | INVITE | INVITE | INVITE | |----------------------------->|-------------------------->| ----------------------->|------------------------------>|------------------------->| | | | | | | | 100 trying | 100 trying | | | | | <-------------------------- |<------------------------- | | | | | 180 Ringing | 180 Ringing | 180 Ringing | 180 Ringing | 180 Ringing | | <--------------------------- | <----------------------- |<---------------------- | <---------------------------- | <---------------------- | | | | | | | | 200 OK | 200 OK | 200 OK | 200 OK | 200 OK | | <--------------------------- | <---------------------- | <---------------------- | <---------------------------- | <---------------------- | | | | | | | | ACK | ACK | ACK | | | | -----------------> | -----------------------> | ----------------------> | ?????? | | Set up: user 1: Internal IP: 192.x.x.3 external ip: 107.x.x.78 user 2: Internal IP: 192.x.x.3 external ip: 107.x.y.249 Nat routers are FC5 linux PCs on which I have enabled SNAT and posrt forwarding. In above case user2 is not getting ACK. fix_nated_contact() API of SER is modifying contact header to the destination ip of receiving message. e.g. SER proxy receives REGISTER meassage from user 1 from NAT router 1 then it changing contact ip to the dst ip of REGISTER message i.e. Nated ip of user1. Because of this RFEGISTER and INVITE goes fine. But in case of 200 OK of INVITE comming from user 2 dst ip is IP of SIP server, so SER adds server ip in the contact header of 200Ok because of this user1 is sending ACK to SERVER but not to user2. If I ignore the ACK (just for testing) and start media using RTPPROXY then media packets are going to SERVER but not to user2 because force_rtp_proxy follows the same principle like fix_nated_contact(). Please provide me some pointers. I am not able to unserstand how use1 will know that 200OK is came from user 2. In 200OK message there no reference of the nated IP of user2 below is my SER config file. # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=6 fork=no log_stderror=yes listen=107.108.70.201 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" #unix_sock="/tmp/ser_sock" 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/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" #loadmodule "/usr/local/lib/ser/modules/domain.so" #loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("auth_db", "calculate_ha1", 1) #modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling # if (method == "INVITE") { # fix_nated_sdp("1"); # Add direction=active to SDP # }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { # if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; # } else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; # }; # consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); # if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; # }; # if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; # }; # consume_credentials(); if (!save("location")) { sl_reply_error(); }; } Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/

2 1

[Users] openser[3844]: ERROR:tm:write_to_unixsock: writev failed: Message too long
by laurent schweizer 15 Apr '07

15 Apr '07

Hi with openser 1.1.1 I have this error Apr 15 10:27:18 byll4 openser[3844]: ERROR: tsend_datagram_ev: failed to send: (90) Message too long Apr 15 10:27:18 byll4 openser[3844]: ERROR:tm:write_to_unixsock: writev failed: Message too long Apr 15 10:27:18 byll4 openser[3844]: ERROR:tm:t_write_unix: write_to_unixsock failed Laurent # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes # yes log_stderror=no #yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 unix_sock="/tmp/ser_sock" listen=udp:195.129.125.57:5060 # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/rr/rr.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/textops/textops.so" #loadmodule "modules/avp/avp.so" loadmodule "modules/avpops/avpops.so" loadmodule "modules/xlog/xlog.so" #loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- modparam("xlog", "buf_size", 8192) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam( "avpops", "avp_aliases", "billing_data=i:67" ) modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append", "billing_headers: ua=$hdr(User-Agent)") #modparam("tm", "tw_append", "billing_headers: ua=$hdr(User-Agent) ;pbilling=$avp(i:67);x-billingId=$hdr(x-billingId);remote-party-id=$hdr(Remote-Party-ID)") modparam("tm", "fr_timer", 200) # appends for dtmf per INFO #modparam( "tm", "tw_append", # "info_append: ctlength=$hdr(Content-Length); cttype=$hdr(Content-Type); msg=$msg(body)") # main routing logic route{ xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\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"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); return; } sl_send_reply("100", " trying "); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); log("step 1\n"); xlog("L_ERR", "step 1\n"); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { log("loose route\n"); xlog("L_ERR", "loose route\n"); append_hf("P-hint: rr-enforced\r\n"); if (!t_relay()) { sl_reply_error(); } return; }; xlog("L_ERR", "step t_newtran\n"); # make transaction # if (method != "ACK") { if (!t_newtran()){ log("could not create transaction\n"); xlog("L_ERR", "could not create transaction \n"); sl_send_reply("500","could not create transaction"); return; } #} log("SIP message \n"); xlog("L_ERR", "step 2\n"); # actively absorb ACKs if (method == "ACK") { log("ACK ... \n"); xlog("L_ERR", "ACK ...\n"); if (!t_relay()) { sl_reply_error(); } #t_relay(); # break; } # pass INFO to SEMS # if (method=="INFO") { # if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ # log("could not contact sems\n"); # t_reply("500","could not contact media server"); # } # } if (uri =~ "sip:.*@") { xlog("L_ERR", "to sems\n"); # if (method=="INVITE") # { # avp_write("$hdr[x-billingId]","$billing_data"); #} if (!t_write_unix("/tmp/sems_sock","billing/billing_headers")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } t_reply("404","Not found in ser .... 1");

2 1

[Users] using LCR problem
by Jayesh Nambiar 15 Apr '07

15 Apr '07

Hi everyone, I am using the LCR module with almost 45000 rows in the LCR table. My server processes a huge call volumes. I thought everything was working fine but I started getting complains of very low connects. Then I observed that Openser was taking time to process calls and when I just unloaded the LCR module from the script, everything worked fine and openser could process all calls properly. I had made the following changes in the config.h and lcr_mod.c file: Shared memory = 256 PKG_Memory = 4*1024*1024 MAX_NO_LCRS=50000 I really need the functionality of LCR in my script for cost efficiency purposes. The server also has 4GB of memory but something in Openser seems to be wrong !!! I also upgraded from OSer 1.1.1 to 1.2 but still same problem :( Can someone please help me out with this? Thanks, w/regards, Jayesh Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/

2 1

[Serusers] Respond to INVITE message for SER
by Goh, Tee-Yong 14 Apr '07

14 Apr '07

Dear all, May i seek your advice. My INVITE message sending to the SER is not successful with error "Status: 500". The SER does not respond back a 200 OK back to the client. Please kindly advice if anyone of you have the solution or encounter such problem before? Thanks. Regards, Howard This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

4 5

RE: [Serusers] SIP SER
by Goh, Tee-Yong 14 Apr '07

14 Apr '07

Hi Karim, Yes you are right. Please advice. Thank you. Regards, Howard -----Original Message----- From: karim basraoui [mailto:basraouik@gmail.com] Sent: Saturday, March 03, 2007 8:01 PM To: Goh, Tee-Yong Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER So if i understand, when a clint registers, SER will automatically send invites to the other clients and make connections with them? 2007/3/3, Goh, Tee-Yong < <mailto:gohty@m1.com.sg> gohty(a)m1.com.sg>: Dear Karim, Thank you for your reply. Yes you are right. I have having difficulty searching for this for past 2 months. Yes. Here is what i want. Once i receive the REGISTER messages, my SER will send out "INVITE" message to my other clients. This is what i am intended to do. However, i could not search for the right coding inside the ser-0.9.6. Please advice and apologies if any inconvenience caused for you. Thanks. Regards, Howard -----Original Message----- From: karim basraoui [mailto: basraouik(a)gmail.com] Sent: Saturday, March 03, 2007 7:18 PM To: Goh, Tee-Yong Subject: Re: [Serusers] SIP SER Hi, Do you want to make changes on SER behaviour when it receives REGISTER messages. If it's the case you can make a condition in the "ser.cfg" file, in the bloc "route{ ... }". for example: route { if (method == "REGISTER") { #what you want to do}; } 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg <mailto:gohty@m1.com.sg> >: Hi, May i know from the SER-0.9.6, where can i look for and make changes for the SER when receiving "REGISTER" message and responsing " 200 OK" back to the corresponding node or clients? Would appreciate your assistance and advice as soon as possible as i have difficultly locating the coding. Please advice where i can find the script to edit the above mentioned messages. Apologies for the inconvenience caused. Thank you. Regards, Howard _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer <http://m1.com.sg/M1/misc/disclaimer> _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer <http://m1.com.sg/M1/misc/disclaimer> _____ -- BASRAOUI Karim www.basraouik.afrikart.net This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

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[Users] SEMS: ivr API
by Helmut Kuper 14 Apr '07

14 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have some trouble to get redirect.py running as an ivr module, cause ivr module's api has changed. Unfortunately I wasn#t able to find any new documentaion about sems' ivr except some hints that documentation isn't actual. Does anybody has the documented ivr API of sems 0.10? Thanks in advance Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGH4QM4tZeNddg3dwRAhycAJ97h/WkjOGAmzr/KbmSnB0QNs8DngCfboEN +qKBWEbg95HlLMq55hPhbEE= =jMhC -----END PGP SIGNATURE-----

2 1

[Serusers] SER, SEMS voicemail and
by Meidiana Ten 13 Apr '07

13 Apr '07

Hi there, I'm sorry for bothering all this time with same questions I still can't solve the problem because nobody reply it. I really don't have any idea about how SEMS works. I setup eyebeam to forward call to voicemail number which is 115@domain for more info, the voicemail address was one of the user in subscriber/aliases table of SER that I added. When the call forwarded, SER have to check to usrloc for the address. It automatically appear that User not found on the softphone. Will it be any way that I don't have to create the address for voicemail address but SER can forward the call for SEMS? Or maybe my softphone forward call setting isn't right?Btw, I'm using eyebeam. I hope that someone reply for my mail this time. I'm sorry if my english is bad. Regards, Meidiana --------------------------------- Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta.

2 4

[Users] NEW FEATURE: support for multiple RTPproxies
by Bogdan-Andrei Iancu 13 Apr '07

13 Apr '07

Hi, thanks to Ancuta Onofrei, a new member of Voice System's team, NATHELPER module now supports multiple rtpproxies - actually you can define multiple sets of rtpproxies - nathelper performs load-balancing over a set and the user can select the set. For more details, see: http://www.openser.org/docs/modules/devel/nathelper.html#AEN76 The definition of the rtpproxy socket was extended to allow definition of multiple sets. See: http://www.openser.org/docs/modules/devel/nathelper.html#AEN140 Also, a new function was added to set from script what set of rtpproxies is to be used. See: http://www.openser.org/docs/modules/devel/nathelper.html#AEN298 Example: # multiple sets of multiple rtproxies modparam("nathelper", "rtpproxy_sock", "1 == udp:localhost:12221 udp:localhost:12222") modparam("nathelper", "rtpproxy_sock", "2 == udp:localhost:12225") ..... set_rtp_proxy_set("2"); force_rtp_proxy(); ..... regards, bogdan

1 0

[Users] Module tm, function t_lookup_request
by Wolfgang Hottgenroth 13 Apr '07

13 Apr '07

Hi, what happens to the t_lookup_request function of the tm module in the 1.2.0 release? I was using this function in my openser script and no it disappears. Although the underlying t_check function still exists. What is the reason? Thanks, Wolfgang

2 6

[Users] Why do not have 407 Proxy Authentication Required
by Mike Lee 13 Apr '07

13 Apr '07

Dear friends: I have installed openser-1.1.0-tls version, and it works fine with mysql, the sip flow is below: UA OpenSER invite----------------> <----------------100 trying <----------------101 dialog establishement <----------------180 ringing ..... My question is, why not have the 407 Proxy Authentication Required in the per call, see my openser.cfg file: # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu 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/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/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" # ----------------- 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{ # 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("localhost", "subscriber")) { www_challenge("localhost", "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; } ============================ thanks. Mike --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

3 2

[Serusers] using SER
by gabrielyans 13 Apr '07

13 Apr '07

Hi I was looking for a softswitch and found that SER could be usefull for what I am going to implement. I want to have a Signalling GW (I think it will be SER) who will receive calls and then direct them according to his routing table. SER will transfer calls to Asterisk, and then Asterisk (together with A2billing) will make the accounting for that call. Is it the right choise to use SER? Actually, I cannot find any forum concerning SER to check if there are already someone who is using SER as I am planning to use. Also, concerning WEB interface - it seams that SERWeb gives only user-management interface and not administrator interface - am I right? Please advise. gabrielyans

3 2

[Serusers] SER + SEMS problem with ACK
by laurent schweizer 13 Apr '07

13 Apr '07

Hello, I'm using sems with ser 0.9.7-pre6 and I have a problem with ACK, as you can see , the other part is not acceting them, what is wrong... ? just, the SIP is going via a IPSEC vpn. Thanks for your help Laurent ? 212.190.89.140 -> 212.249.14.236 ............s....Y...5..E.....@....d..Y...}9.......fSIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch=z9hG4bK8ec6.3148d226.0..From:417875185 51 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-ev ent..Content-Length: 0.... U 212.190.89.137:5060 -> 195.129.125.57:5060 SIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch= z9hG4bK8ec6.3148d226.0..From: 41787518551 <sip:+41787518551@bylltel.com >;tag=568D320F-461F4 5CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40 807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-event..Content-Length: 0.... ? 212.249.14.236 -> 212.190.89.140 ................koG.B...E.....@.@..&..}9..Y.........ACK sip:+41787518551@212.190.89.137 SIP/2.0..Via: SIP/2.0/UDP 195.129.125.57 ;branch=z9hG4bK8ec6.3148d 226.0..From: 41787518551 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960(a)212.249.14.23 6..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..CSeq: 10 ACK..User-Agent: Sip EXpress router(0.9.7-pre6(x86_64/linux))..Content-Length: 0... . ? 212.190.89.140 -> 212.249.14.236 ................gT.[_...E.....@....c..Y...}9.......fSIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch=z9hG4bK8ec6.3148d226.0..From:417875185 51 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-ev ent..Content-Length: 0.... U 212.190.89.137:5060 -> 195.129.125.57:5060 SIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch= z9hG4bK8ec6.3148d226.0..From: 41787518551 <sip:+41787518551@bylltel.com >;tag=568D320F-461F4 5CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40 807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-event..Content-Length: 0.... ? 212.249.14.236 -> 212.190.89.140 ............|ovW. ..5.izE.....@.@..&..}9..Y.........ACK sip:+41787518551@212.190.89.137 SIP/2.0..Via: SIP/2.0/UDP 195.129.125.57 ;branch=z9hG4bK8ec6.3148d 226.0..From: 41787518551 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960(a)212.249.14.23 6..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..CSeq: 10 ACK..User-Agent: Sip EXpress router(0.9.7-pre6(x86_64/linux))..Content-Length: 0...

1 0

[Serusers] Avpops'exported function
by chungyu 13 Apr '07

13 Apr '07

Dear all: avpops module' exported function avp_db_delete() can do delete one record? example: I use "test" as default table that has two fields. username | number ------------------- bob | 1 alice | 1 bob | 2 my configuration: modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "test") modparam("avpops", "username_column", "username") modparam("avpops", "avp_aliases", "num=i:55") modparam("avpops", "db_scheme", "number_scheme:table=test;value_col=number;value_type=string") if (method=="INVITE"){ avp_db_load("$from/username", "$num/$number_scheme"); //load from table "test" ,number column as string value into AVP name '$num' avp_db_delete("$from/username", "s:test") //delete all bob' record }; if bob calling then bob'record has deleted username | number -------------------- alice | 1 But i want delete a record. ex: delete number value is 1 for bob username | number -------------------- alice | 1 --------------------- bob | 2 avp_db_delete() can do delete one record? Another question is avp_db_store() can do insert new data in table? ex: insert into test (username,number) values ("mary","3") username | number ------------------- alice | 1 ------------------- bob | 2 ------------------- mary | 3 avp_db_store() can do it? sorry,I still can't understand what it mean about avp_db_store() and avp_db_delete(). Can tell me? Thanks a lot in advance Chungyu

3 3

[Serusers] fifo
by Javier 12 Apr '07

12 Apr '07

Hi, I have a problem, I use the ser 0.8.14, and I update to 0.10.99 dev 30 I have a module that send BYE to all part of comunication using fifo. but this don't work in 0.10.99 version can you help me ? best regards Lou __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

2 1

[Users] MOH SIP UA - does such a program exist?
by Aaron Daubman 12 Apr '07

12 Apr '07

Greetings, I'm looking for a (IPv6 capable) SIP UA that functions basically as a Music-on-Hold type client. This will be used in a lab, and the basic desired functionality is to have this UA register its extension, and then have any of our soft/hard-phones be able to dial the extension and hear the music playing (WAV, MP3, whatever) - note that there is no need for it to be able to handle simultaneous calls... a SIP softphone that allowed auto-answer and auto-wav-playback would be exactly what I'm looking for... I really just want to remove the need to have a live person on the other end of a call to answer and speak to demonstrate VoIP functionality during lab demonstrations and quality experiments... Thanks, ~Aaron

3 2

[Serusers] Parsing X-Remote-Info header using exec
by Graham Lewis 12 Apr '07

12 Apr '07

Some INVITE messages to my SER include an additional header "X-Remote-Info" that I want to be able to parse and pass to an external program using exec_msg What is the best way to go about this? The well known headers are passed as ENV variables but other than re-writing the module to include a SIP_X-REMOTE-INFO how can I pass the entire INVITE to an external script? TIA -- Graham Lewis ---- Msg sent via Spitfire @Mail - http://www.spitfire.co.uk/

2 1

[Serusers] Auto start at boot up
by POON Ben 12 Apr '07

12 Apr '07

Hi all, I was wondering if I can auto start SER at boot up? I am running Solaris 10 with SER 0.9.6. I've done this: ln /usr/local/sbin/serctl /etc/rc3.d/S99ser ln /usr/local/sbin/serctl /etc/rc0.d/K00ser ln /usr/local/sbin/serctl /etc/rc1.d/K00ser ln /usr/local/sbin/serctl /etc/rc2.d/K00ser ln /usr/local/sbin/serctl /etc/rc3.d/K00ser But after reboot it still doesn't start... Did i make the links wrong? Or is there another setting somewhere? Thanks Ben

3 4

[Users] presence in openser 1.2 ... cannot insert row?
by Cesc 12 Apr '07

12 Apr '07

Hi, I am testing the presence in openser1.2 ... running a ser.0.9.6 as a registrar/proxy which forwards all presence messages to openser. As a client, I use xlite 3.0 ... I had an old semi-functional openser working, pre-1.2 ... and now I upgraded to 1.2 from svn and it ain't working :) The config file I modified, as well as creating the DBTEXT files (all) from scratch using the textdb.sh script. Now, you can see the attached config file ... and the whireshark trace with the subscribe and publish messages received ... What happens can be seen here, in the log I get ... most interesting are the lines with ERROR written on them ;) They tell that a new row could not be inserted ... and that some lookup failed on the database ... I guess a familiar eye can catch that easy. 4(2517) parse_headers: flags=78 4(2517) t_lookup_request: start searching: hash=50044, isACK=0 4(2517) DEBUG: RFC3261 transaction matching failed 4(2517) DEBUG: t_lookup_request: no transaction found 4(2517) ============================================================================================= 4(2517) The request method used is to [SUBSCRIBE] information. (handle subscribe) 4(2517) ============================================================================================= 1(2513) SIP Request: 1(2513) method: <SUBSCRIBE> 1(2513) uri: <sip:Chelsea@cafes.org> 1(2513) version: <SIP/2.0> 1(2513) parse_headers: flags=2 1(2513) Found param type 232, <branch> = <z9hG4bKa4e7.bd4e2945.0>; state=16 1(2513) end of header reached, state=5 1(2513) parse_headers: Via found, flags=2 1(2513) parse_headers: this is the first via 1(2513) After parse_msg... 1(2513) preparing to run routing scripts... 1(2513) parse_headers: flags=100 1(2513) Found param type 232, <branch> = <z9hG4bK-d87543-2641a76387057623-1--d87543->; state=6 1(2513) Found param type 235, <rport> = <4326>; state=16 1(2513) end of header reached, state=5 1(2513) parse_headers: Via found, flags=100 1(2513) parse_headers: this is the second via 1(2513) DEBUG:maxfwd:is_maxfwd_present: value = 16 1(2513) ============================================================================================== 1(2513) parse_headers: flags=10 1(2513) DEBUG:parse_to:end of header reached, state=10 1(2513) DBUG:parse_to: display={}, ruri={sip:Chelsea@cafes.org} 1(2513) DEBUG: get_hdr_field: <To> [25]; uri=[sip:Chelsea@cafes.org] 1(2513) DEBUG: to body [<sip:Chelsea@cafes.org> ] 1(2513) DEBUG: add_param: tag=1753b200 1(2513) DEBUG:parse_to:end of header reached, state=29 1(2513) DBUG:parse_to: display={"X-lite user 1"}, ruri={sip:Klaas@cafes.org} 1(2513) New Message received: [7] from [sip:Klaas@cafes.org] 1(2513) Request method used:[SUBSCRIBE]. 1(2513) ============================================================================================= 1(2513) DEBUG: t_newtran: T on entrance=0xffffffff 1(2513) parse_headers: flags=ffffffffffffffff 1(2513) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 1(2513) DEBUG: get_hdr_body : content_length=0 1(2513) found end of header 1(2513) parse_headers: flags=78 1(2513) t_lookup_request: start searching: hash=32330, isACK=0 1(2513) DEBUG: RFC3261 transaction matching failed 1(2513) DEBUG: t_lookup_request: no transaction found 1(2513) ============================================================================================= 1(2513) The request method used is to [SUBSCRIBE] information. (handle subscribe) 1(2513) ============================================================================================= 1(2513) parse_headers: flags=ffffffffffffffff 1(2513) PRESENCE: handle_subscribe: 'expires' found 1(2513) PRESENCE: handle_subscribe: lexpire= 3600 1(2513) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: <sip:Chelsea@cafes.org> 1(2513) PRESENCE:handle_subscribe: generating to_tag 1(2513) PRESENCE :proces_rr: out rr [<sip:192.168.1.150;ftag=1753b200;lr=on>] 1(2513) DBT:dbt_query: new res with 2 cols 1(2513) DBT:dbt_result_new: new res with 2 cols 1(2513) PRESENCE:handle_subscribe:The query in table watches returned no result 1(2513) DBT:dbt_table_check_row: NULL value not allowed - field 0 1(2513) DBT:db_insert: cannot insert the new row!! 1(2513) DBT:db_insert: make clean! 1(2513) PRESENCE:handle_subscribe: ERROR while inserting into watchers table 1(2513) PRESENCE:handle_subscribe: ERROR occured 1(2513) DEBUG: cleanup_uac_timers: RETR/FR timers reset 1(2513) DEBUG: add_to_tail_of_timer[2]: 0xb5d79c78 (45) 1(2513) DEBUG:tm:UNREF_UNSAFE: after is 0 1(2513) DEBUG:destroy_avp_list: destroying list (nil) 1(2513) receive_msg: cleaning up 2(2515) SIP Request: 2(2515) method: <PUBLISH> 2(2515) uri: <sip:Klaas@cafes.org> 2(2515) version: <SIP/2.0> 2(2515) parse_headers: flags=2 2(2515) Found param type 232, <branch> = <z9hG4bK5aa3.0e4b2fb7.0>; state=16 2(2515) end of header reached, state=5 2(2515) parse_headers: Via found, flags=2 2(2515) parse_headers: this is the first via 2(2515) After parse_msg... 2(2515) preparing to run routing scripts... 2(2515) parse_headers: flags=100 2(2515) Found param type 232, <branch> = <z9hG4bK-d87543-7441d6073759ed7a-1--d87543->; state=6 2(2515) Found param type 235, <rport> = <4326>; state=16 2(2515) end of header reached, state=5 2(2515) parse_headers: Via found, flags=100 2(2515) parse_headers: this is the second via 2(2515) DEBUG:maxfwd:is_maxfwd_present: value = 16 2(2515) ============================================================================================== 2(2515) parse_headers: flags=10 2(2515) DEBUG:parse_to:end of header reached, state=10 2(2515) DBUG:parse_to: display={"X-lite user 1"}, ruri={sip:Klaas@cafes.org} 2(2515) DEBUG: get_hdr_field: <To> [38]; uri=[sip:Klaas@cafes.org] 2(2515) DEBUG: to body ["X-lite user 1"<sip:Klaas@cafes.org> ] 2(2515) DEBUG: add_param: tag=eb194522 2(2515) DEBUG:parse_to:end of header reached, state=29 2(2515) DBUG:parse_to: display={"X-lite user 1"}, ruri={sip:Klaas@cafes.org} 2(2515) New Message received: [5] from [sip:Klaas@cafes.org] 2(2515) Request method used:[PUBLISH]. 2(2515) ============================================================================================= 2(2515) DEBUG: t_newtran: T on entrance=0xffffffff 2(2515) parse_headers: flags=ffffffffffffffff 2(2515) get_hdr_field: cseq <CSeq>: <1> <PUBLISH> 2(2515) DEBUG: get_hdr_body : content_length=439 2(2515) found end of header 2(2515) parse_headers: flags=78 2(2515) t_lookup_request: start searching: hash=15013, isACK=0 2(2515) DEBUG: RFC3261 transaction matching failed 2(2515) DEBUG: t_lookup_request: no transaction found 2(2515) ============================================================================================= 2(2515) The request method used is to [PUBLISH] information. (handle publish) 2(2515) ============================================================================================= 2(2515) parse_headers: flags=ffffffffffffffff 2(2515) PRESENCE:handle_publish: SIP-If-Match not found 2(2515) PRESENCE: generate_ETag: etag= a.1175401932.2515.3 / 19 2(2515) PRESENCE:handle_publish: new etag = a.1175401932.2515.3 2(2515) PRESENCE: handle_publish: 'expires' found 2(2515) PRESENCE: handle_publish: lexpire= 3600 2(2515) PRESENCE: handle_publish: 'To' header ALREADY PARSED: <sip:Klaas@cafes.org> 2(2515) parse_headers: flags=ffffffffffffffff 2(2515) PRESENCE:update_presentity: inserting 7 cols intotable 2(2515) DBT:dbt_table_check_row: NULL value not allowed - field 0 2(2515) DBT:db_insert: cannot insert the new row!! 2(2515) DBT:db_insert: make clean! 2(2515) PRESENCE:update_presentity: Error while inserting new presentity 2(2515) PRESENCE:update_presentity: ERROR occured 2(2515) PRESENCE:handle_publish: ERROR occured while updating presentity 2(2515) PRESENCE: handle_publish: ERROR occured 2(2515) DEBUG: cleanup_uac_timers: RETR/FR timers reset 2(2515) DEBUG: add_to_tail_of_timer[2]: 0xb5d77d08 (45) 2(2515) DEBUG:tm:UNREF_UNSAFE: after is 0 2(2515) DEBUG:destroy_avp_list: destroying list (nil) 2(2515) receive_msg: cleaning up 4(2517) parse_headers: flags=ffffffffffffffff 4(2517) PRESENCE: handle_subscribe: 'expires' found 4(2517) PRESENCE: handle_subscribe: lexpire= 3600 4(2517) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: <sip:Klaas@cafes.org> 4(2517) PRESENCE:handle_subscribe: generating to_tag 4(2517) PRESENCE :proces_rr: out rr [<sip:192.168.1.150;ftag=e7740874;lr=on>] 4(2517) [p_user]= Klaas [p_domain]= cafes.org [w_user]= Klaas [w_domain]= cafes.org 4(2517) [event]= presence.winfo [staus]= active [expires]= 3600 4(2517) [to_tag]= 10.2517.1175401973.2 [from_tag]= e7740874 4(2517) PRESENCE: update_subscription ... 4(2517) [p_user]= Klaas [p_domain]= cafes.org [w_user]= Klaas [w_domain]= cafes.org 4(2517) [event]= presence.winfo [staus]= active [expires]= 3600 4(2517) [to_tag]= 10.2517.1175401973.2 [from_tag]= e7740874 4(2517) expires: 3600 4(2517) PRESENCE:update_subscription:Inserting into database: n_query_cols:14 4(2517) [0] = to_user Klaas 4(2517) [1] = to_domain cafes.org 4(2517) [2] = from_user Klaas 4(2517) [3] = from_domain cafes.org 4(2517) [4] = event presence.winfo 4(2517) [5] = event_id 4(2517) [6] = callid MGUwMzgyNGFiMWE5ZGIxZjg1NGQwNDI5N2JlOTFmYjA. 4(2517) [7] = to_tag 10.2517.1175401973.2 4(2517) [8] = from_tag e7740874 4(2517) [9] = contact sip:Klaas@192.168.3.172:4326 4(2517) [10] = status active 4(2517) [11] = cseq 1 4(2517) DBT:dbt_table_check_row: NULL value not allowed - field 0 4(2517) DBT:db_insert: cannot insert the new row!! 4(2517) DBT:db_insert: make clean! 4(2517) PRESENCE:update_subscription: ERROR while storing new subscription 4(2517) PRESENCE:update_presentity: ERROR occured 4(2517) PRESENCE:handle_subscribe: ERROR while updating database 4(2517) PRESENCE:handle_subscribe: ERROR occured 4(2517) DEBUG: cleanup_uac_timers: RETR/FR timers reset 4(2517) DEBUG: add_to_tail_of_timer[2]: 0xb5d763b8 (45) 4(2517) DEBUG:tm:UNREF_UNSAFE: after is 0 4(2517) DEBUG:destroy_avp_list: destroying list (nil) 4(2517) receive_msg: cleaning up 3(2516) 1(2513) SIP Request: 1(2513) method: <SUBSCRIBE> 1(2513) uri: <sip:Chelsea@cafes.org> 1(2513) version: <SIP/2.0> 1(2513) parse_headers: flags=2 1(2513) Found param type 232, <branch> = <z9hG4bKa4e7.bd4e2945.0>; state=16 1(2513) end of header reached, state=5 1(2513) parse_headers: Via found, flags=2 1(2513) parse_headers: this is the first via 1(2513) After parse_msg... 1(2513) preparing to run routing scripts... 1(2513) parse_headers: flags=100 1(2513) Found param type 232, <branch> = <z9hG4bK-d87543-2641a76387057623-1--d87543->; state=6 1(2513) Found param type 235, <rport> = <4326>; state=16 1(2513) end of header reached, state=5 1(2513) parse_headers: Via found, flags=100 1(2513) parse_headers: this is the second via 1(2513) DEBUG:maxfwd:is_maxfwd_present: value = 16 1(2513) ==============================================================================================

2 1

[Users] Avpops'exported function
by chungyu 12 Apr '07

12 Apr '07

Dear all: Avpops module' exported function avp_db_delete() can do delete one record? example: I use "test" as default table that has two fields. username | number ------------------- bob | 1 alice | 1 bob | 2 my configuration: modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "test") modparam("avpops", "username_column", "username") modparam("avpops", "avp_aliases", "num=i:55") modparam("avpops", "db_scheme", "number_scheme:table=test;value_col=number;value_type=string") if (method=="INVITE"){ avp_db_load("$from/username", "$num/$number_scheme"); //load from table "test" ,number column as string value into AVP name '$num' avp_db_delete("$from/username", "s:test") //delete all bob' record }; if bob calling then bob'record has deleted username | number -------------------- alice | 1 But i want delete a record. ex: delete number value is 1 for bob username | number -------------------- alice | 1 --------------------- bob | 2 avp_db_delete() can do delete one record? Another question is avp_db_store() can do insert new data in table? ex: insert into test (username,number) values ("mary","3") username | number ------------------- alice | 1 ------------------- bob | 2 ------------------- mary | 3 avp_db_store() can do it? sorry,I still can't understand what it mean about avp_db_store() and avp_db_delete(). Can tell me? Thanks a lot in advance Chungyu

4 4

[Users] un-asked-for CANCEL being generated
by Mark Price 12 Apr '07

12 Apr '07

I'm using openser 1.1.x, and seeing something rather strange. We are using openser as a proxy with openser and asterisk on box A, asterisk on box B. Openser listens on ports 5060 and 5065. Asterisk listens on port 5070 on box A and 5060 on box B. When a call arrives from PSTN, it works fine if I forward to asterisk in box B. If I forward to asterisk on box A, everything seems to be going well and then openser generates a CANCEL to the receiving phone for seemingly no good reason. You can see from the packet below that the cancel is being generated by openser, because of the user agent. What might cause this? Thanks, Mark Price U 66.129.95.24:5065 -> 71.199.89.40:16742 CANCEL sip:9043067833@71.199.89.40:16742;rinstance=f5f4de634aa6bff6;transport=udp SIP/2.0..Via: SIP/2.0/UDP 66.129.95.24:5065;branch=z9hG4bK1c31.2cb72f11.0..From: "Mark Price" <sip:9046268172@66.129.95.5>;tag=as11c24889..Call-ID: 1a095b0e1901cead54e8d68c63271aad@66.129.95.5..To: <sip:9043067833@joinuneta.com:5065>..CSeq: 102 CANCEL..User-Agen t: OpenSer (1.1.0-tls (i386/linux))..Content-Length: 0....

2 4

[Users] OpenSER + RTPProxy as Session Border Controller (SBC)
by Martin Klisch 12 Apr '07

12 Apr '07

Hi, is it possible to use OpenSER + RTProxy as a small Session Border Controller? I want to pass ALL calls to an extra OpenSER+RTPProxy, so ALL calls use RTP Sessions via RTPProxy. Can the RTP Proxy signal to the SIP Proxy, if a RTP Session timeouts, so the SIP Proxy can signal a BYE to both legs of the call? I need something like this, to not account timeouted calls. For example broken calls, which never end. any other ideas? professional session border controllers are very very very expensive.

2 1

[Users] TLS and asterisk ?
by Bodin Bruno 12 Apr '07

12 Apr '07

Hi, I know asterisk doesn't support tls but that something strange in SIP protocol : TLS UA -> Openser -> no tls UA : that work TLS UA -> Openser -> asterisk : that don't work any one know something about this ? to use asterisk I make rewritehostport. thank for help

3 8

[Users] How to change the expire parameter in a reply message of REGISTER?
by 杨冬 12 Apr '07

12 Apr '07

hi, all: I want to make a client to send the REGISTER update message in a short time, which is used to prevent the NAT device to close the port for the client. So I want to rewrite the reply message for REGISTER and change the expire parameter to a value of my setting. How can I achieve this? Thanks. andy.yang 2007/04/10

4 6

[Users] Redirecting number of ISUP-IAM to Diversion field of SIP-INVITE translation
by Kettmann, Peter 12 Apr '07

12 Apr '07

Hi, we are using Asterisk (1.2.12.1), chan_ss7 library (0.8.4) and a Digium E1 card to perform calls between IP and PSTN/GSM networks. In the next step we need to translate Redirecting number from ISUP-IAM message to Diversion field of SIP-INVITE message. Do you have an idea to make it on Asterisk ? Is it perhaps possible via the OpenSer Diversion module? Thanks for your help, Peter

2 1

[Users] Possible to use subst() to alter first line of INVITE?
by Leif Madsen 12 Apr '07

12 Apr '07

Evening all! I'm trying to use subst() to alter the first line of an INVITE I'm sending to a remote Asterisk box, although I have been unable to get this working. Perhaps I need to use a different function? The INVITE I'm working with looks like (IPs have been changed to protect the innocent): INVITE sip:s@192.168.1.250 SIP/2.0 Record-Route: <sip:192.168.1.190;lr;ftag=as62d28c09> Via: SIP/2.0/UDP 192.168.1.190;branch=z9hG4bKcce2.b907e716.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK5a287301;rport=5060 From: "lmadsen2#lmentinc" <sip:lmadsen2#lmentinc@192.168.1.147>;tag=as62d28c09 To: <4166289921@switch1> Contact: <sip:lmadsen2#lmentinc@192.168.1.147> Call-ID: 791d9cc101c5cbdd2896c0d357cf644c(a)192.168.1.147 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 69 Remote-Party-ID: "lmadsen2#lmentinc" <sip:lmadsen2#lmentinc@192.168.1.147>;privacy=off;screen=no Date: Thu, 12 Apr 2007 05:00:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 I have two subst() I've been playing with. This one works which alters the To: line in the INVITE: subst('/^To: (.*)sip:[^@]*+(.*)$/To: \1$hdr(X-DID)\2/g'); This one does not, which I would like to alter the INVITE sip:s@192.168.1.250 SIP/2.0 of the header. This is the very first line in the INVITE. subst('/^INVITE (.*)sip:[^@]*+(.*)$/INVITE sip:$hdr(X-DID)\2/ig'); I tested the regex online as well, and it seems to work. Anything I'm missing? Is there another function I can use to alter the line I want, or am I up poop creek? Thanks! -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk

3 3

re:[Users] Sip Tools
by yanlin 12 Apr '07

12 Apr '07

hi, raviprakash sunkara, http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software search here, maybe you can find something useful. i use SIPp to test performance, it can generate calls in certain sequence. regards, yan lin *** Disclaimer: This message may contain privileged and/or confidential information. If you have received this e-mail in error or are not the intended recipient, you may not use, copy, disseminate or distribute it; do not open any attachments, delete it immediately from your system and notify the sender promptly by e-mail that you have done so. Thank you. ***

2 2

[Serusers] re: Malformed CONTACT field in 302 response
by Jignesh Gandhi 11 Apr '07

11 Apr '07

Hello, I am using SER as a redirect server. I recently posted almost the same question. This time the format of the CONTACT is different. It contains call information. Recently I came across an issue where the 302 sent back by SER is not liked by a soft switch. Particularly , the format of the CONTACT field according to the softswitch.. Here is an excerpt of the 302 reply send back ... Session Initiation Protocol Status-Line: SIP/2.0 302 MovedTemporarily Status-Code: 302 Resent Packet: False Message Header Via:SIP/2.0/UDP 70.167.132.2;branch= z9hG4bK-BroadWorks.70.167.132.2-172.20.20.46V5060-0-476027561-38657848-1175625610577 - From:"7000 Arris-EMTA-SIP"<sip:6782347000@70.167.132.2 ;user=phone>;tag=38657848-1175625610577- SIP Display info: "7000 Arris-EMTA-SIP" SIP from address: sip:6782347000@70.167.132.2 SIP tag: 38657848-1175625610577- To:<sip:+16782349999;csel=noind@172.20.20.46:5060;user=phone>;tag= b27e1a1d33761e85846fc98f5f3a7e58.4baf SIP to address: sip:+16782349999 SIP tag: b27e1a1d33761e85846fc98f5f3a7e58.4baf Call-ID:BW144010577030407-1243907316@70.167.132.2 CSeq:476027561 INVITE Contact: sip:+16782349999;csel=noind@172.20.20.37 :5060;user=phone;transport=udp Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 the softswitch is wanting the contact field with <sip:xxxx@xxx.xxx.xxx.xxx:5060> like a FROM or TO URI. It is claiming that since it contains call information within the sip URI, it should be enclosed withing < >. Here is ser.cfg part that does the sl_send_reply(); # do a stateless redirect, if return code is correct if (method=="INVITE") { xlog("L_INFO", "SourceIP <%is> \n"); xlog("L_INFO", "From-uri<%fu>, r-uri <%ru> \n"); if (!gl_redirect()) << custom function that just gets a Server to redirect to { sl_send_reply("480", "TemporarilyUnavailable"); xlog("L_WARN", "Sending a 480 response with r-uri <%ru>\n"); break; } else { sl_send_reply("302", "MovedTemporarily"); xlog("L_DBG", "Sending a 302 response with r-uri <%ru>\n"); break; } }; Any help is appreciated.** thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com

2 2

[Users] mysql database accounting problems
by Tanguy Bouzéloc 11 Apr '07

11 Apr '07

Hello, I've some issues with my mysql accounting, I can only log successful calls in the acc table and I've no informations about missed calls. In mysql accounting I've only sip 200 ok and no errors. I've not this behavior in my syslog. It's very important for us for evaluate the performance of our providers and we can only do it with a database accouting. For example, If I pass a call and I let it ring til operator timeout, I've no trace of this call in the accounting table. I've also another question, the lcr and gw tables have very small fixed limits (256 for lcr and 32 for gw), since we have much more gateways and sdas I've updated theses limits to a value which fit our needs, do you thinks it can be a problem ? Thanks in advance, Tanguy Bouzéloc Init-Sys # ----------- global configuration parameters ------------------------ #debug=9 fork=yes check_via=no dns=no rev_dns=no port=5060 children=4 sip_warning=0 # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/lcr.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/permissions.so" # ----------------- setting module-specific parameters --------------- modparam("lcr", "db_url", "mysql://???") modparam("lcr", "db_mode", 0) modparam("rr","enable_full_lr",1) modparam("acc", "db_url", "mysql://???") modparam("acc", "db_flag", 1) modparam("acc", "db_localtime", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1 ) modparam("permissions", "db_url", "mysql://???") route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; if (!is_method("REGISTER")) { record_route(); }; if (is_method("BYE") || is_method("CANCEL")) { xlog("L_NOTICE", "End call($ci | $Ri): $fU -> $tU"); if (is_method("BYE")) setflag(1); unforce_rtp_proxy(); }; if (loose_route()) { route(1); return; }; if (!allow_trusted()) { xlog("L_WARN", "Host not trusted ($fu/$si/$rP)"); sl_send_reply("403", "Forbidden"); exit; }; if (!load_gws()) { sl_send_reply("500", "Internal server error, unable to load gateways"); return; } if (!next_gw()) { xlog("L_ERR", "No gw found ($ru/$fu)"); sl_send_reply("503", "Service not available, no gateways found"); return; }; t_on_failure("2"); route(1); } route[1] { setflag(1); if (is_method("INVITE") || is_method("ACK")){ if (is_method("INVITE")) xlog("L_NOTICE", "Incoming call($ci | $Ri): $fU -> $tU"); if (!search("^Content-Length:[ ]*0")) force_rtp_proxy("foc"); # if(src_ip == 217.15.85.178) { # subst('/^From:.*<sip:(.*)@(.*)$/From: "\1" <sip:\1/@\2'); # } t_on_reply("1"); }; if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { setflag(1); if (status=~"[12][0-9][0-9]"){ if (!search("^Content-Length:[ ]*0")) force_rtp_proxy("foc"); }; } failure_route[2] { if (t_check_status("400")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 400 Bad Request"); } else if (t_check_status("401")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 401 Unauthorized"); } else if (t_check_status("403")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 403 Forbidden"); } else if (t_check_status("404")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 404 Not Found"); } else if (t_check_status("405")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 405 Method Not Allowed"); } else if (t_check_status("406")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 406 Not Acceptable"); } else if (t_check_status("407")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 407 Proxy Authentication Required"); } else if (t_check_status("408")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 408 Request Timeout"); } else if (t_check_status("410")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 410 Gone"); } else if (t_check_status("413")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 413 Request Entity Too Large"); } else if (t_check_status("414")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 414 Request-URI Too Long"); } else if (t_check_status("415")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 415 Unsupported Media Type"); } else if (t_check_status("416")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 416 Unsupported URI Scheme"); } else if (t_check_status("420")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 420 Bad Extension"); } else if (t_check_status("421")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 421 Extension Required"); } else if (t_check_status("423")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 423 Interval Too Brief"); } else if (t_check_status("480")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 480 Temporarily Unavailable"); } else if (t_check_status("481")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 481 Call/Transaction Does Not Exist"); } else if (t_check_status("482")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 482 Loop Detected"); } else if (t_check_status("483")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 483 Too Many Hops"); } else if (t_check_status("484")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 484 Address Incomplete"); } else if (t_check_status("485")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 485 Ambiguous"); } else if (t_check_status("486")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 486 Busy Here"); } else if (t_check_status("487")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 487 Request Terminated"); } else if (t_check_status("488")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 488 Not Acceptable Here"); } else if (t_check_status("491")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 491 Request Pending"); } else if (t_check_status("493")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 493 Undecipherable"); } else if (t_check_status("500")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 500 Server Internal Error"); } else if (t_check_status("501")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 501 Not Implemented"); } else if (t_check_status("502")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 502 Bad Gateway"); } else if (t_check_status("503")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 503 Service Unavailable"); } else if (t_check_status("504")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 504 Server Time-out"); } else if (t_check_status("505")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 505 Version Not Supported"); } else if (t_check_status("513")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 513 Message Too Large"); } else { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU Unknown code"); } setflag(1); if (is_method("INVITE") && t_check_status("408|500|503")) { if (!next_gw()) { xlog("L_ERR", "No more gw ($ru/$fu)"); t_reply("486", "Busy"); # t_reply("503", "Service not available"); exit; } t_on_failure("2"); force_rtp_proxy("foc"); t_relay(); } }

4 7

Re: [Users] Unable to dial with openser and UAs (Softphone)
by Tsitsi 11 Apr '07

11 Apr '07

Hi Andreas, Thank for your reply. Yes, this is a sipwise config. Openser : I use Debian Sarge apt repository : deb http://www.openser.org/debian sarge main Mysql 4.1 I have set up all tables building with openser_mysql Softphone: Windows XP and X-lite 3.0 build 34025 Regards, Tsitsi --------------------------------- Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses.

2 3

[Users] enabling ipv6
by shruu thii 11 Apr '07

11 Apr '07

hi all, i'm using fedora 4.i've installed openser 1.0.1.how do i enable this for ipv6? i want to use linphone clients.this is for pc-pc testing on lan. --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

3 5

[Users] Sip Tools
by raviprakash sunkara 11 Apr '07

11 Apr '07

Hello Urs, Anybody can tell , is there to Test load balance in OpenSER and Asterisk , Not Ethereal and ngrep or sniffers. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

3 4

[Users] Openser on ucLinux
by Asterisk Expert 11 Apr '07

11 Apr '07

Hello Everyone, I would like to know if anyone has compiled and installed OpenSER on ucLinux. If yes, can you provide the hardware details and procedure if its different than normal openser installation. -- Thanks Ruchir Brahmbhatt Ecosmob Technologies

1 0

[Users] Cisco/OpenSER called number issues after ENUM lookup
by Luca Corti 11 Apr '07

11 Apr '07

Hello, I've got an AS5350 receiving calls on ISDN E1 and forwarding calls to OpenSER. At the moment ENUM translations are performed on the AS5350 and calls received by OpenSER already contain the correct SIP URIs. Unfortunately I verified that the called party number information is lost in the process and at the called side I cannot distinguish numbers if I associate multiple PSTN numbers to the same sip user. I guess this is Cisco's fault, since it should add a Remote-Party-ID of type "called" to the INVITE. Do I have to move ENUM mapping to OpenSER or is there a quicker solution? thanks Luca

2 2

Re:[Users] how to handle 302-moved temporarily
by Bogdan-Andrei Iancu 11 Apr '07

11 Apr '07

Hi Zahid, a cause might be a bogus received ACK (or improper handled), making openser to retransmit the final response. Please send me privately the openser debug output (use debug 9) for this scenario. regards, bogdan zm23 wrote: >> >> Hi Bogdan, >> I do see an ack sent back from the gw, however, when new >> invite is sent openser relays it according to lcr rules but then a >> 302 is sent back to the gateway. >> >> >> Would it be better to detect and process the 302 message in openser >> itself? >> >> Thanks. >> >> -- >> >> >> >> >> On Mar 30, 2007, at 9:49 AM, Bogdan-Andrei Iancu wrote: >> >>> Hi Zahid, >>> >>> is the second 302 identical to the fist? is there any ACK from CISCO >>> GW to openser as response to first 302? >>> >>> regards, >>> bogdan >>> >>> zm23 wrote: >>>> I made a mistake saying it was a REFER method. This is happening >>>> with just a 302 reply messages. >>>> Any suggestions? >>>> >>>> Thanks. >>>> >>>> >>>> On Mar 28, 2007, at 12:17 PM, zm23 wrote: >>>> >>>>> Hello All, >>>>> I am trying to find out the proper/best way to handle >>>>> REFERs coming from phone. My setup is as follows: >>>>> >>>>> >>>>> Cisco gw openser >>>>> phone gw 2 >>>>> >>>>> >>>>> DID >>>>> 1234 -----> Invite -----------------> >>>>> --------------------> >>>>> >>>>> >>>>> >>>>> <----- 302 msg <--------------------- >>>>> <--------------------302 moved.... >>>>> >>>>> new #5678 sent back >>>>> >>>>> After receiving new invite, openser sends it out to another gateway >>>>> >>>>> 5678 ----> invite ----------------------> >>>>> ---------------------------------------------------> >>>>> >>>>> yet i am still seeing the 302 messages sent back from proxy to >>>>> cisco gw. >>>>> >>>>> <------------------------------- -- 302 moved >>>>> >>>>> >>>>> Call does connect to 5678 but is dropped after few seconds with >>>>> only one way audio. >>>>> >>>>> >>>>> What needs to be done so that when second invite hits proxy, there >>>>> are no more 302s sent back? >>>>> >>>>> Should I process the 302 in proxy itself? >>>>> >>>>> What is the best way to make the 2nd call to 5678 so that it can >>>>> be identified as a separate call? >>>>> >>>>> >>>>> >>>>> >>>>> Thank you in advance for your help in this regards. >>>>> >>>>> --Zahid

1 0

[Users] Does openser support SMS over IMS
by hu li 11 Apr '07

11 Apr '07

Hi, I want to setup an test environment for Short Message Service over IMS network, can openser simulate the IMS network, then I can send short message between two phone through openser in IMS domain? Can openser only be installed in linux OS, not in windows OS? I did not find the binary files for installing in windows. Thanks in advance!

2 1

[Users] XCAP Lite 0.9 released
by Adrian Georgescu 11 Apr '07

11 Apr '07

Change log xcap-lite (0.9) unstable; urgency=low * Validate presrules against presence-rules.xsd * Use Xpath for node selection * Enable multiple domains (embed Domain=example.com in XCAP uri) The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.9.tar.gz Regards, Adrian

1 0

[Users] Debian package of Openser 1.2.0 with TLS compiled in?
by Kerker Staffan 11 Apr '07

11 Apr '07

Hi are there any debian packages released with the TLS support compiled in? i found all other packages for debian, but none with TLS support. best regards /Staffan Kerker --- Staffan Kerker, Saab Communication Ljungadalsgatan 2, 35180 Växjö, Sweden p. +46 470 42185 c. +46 705 391365 m. staffan.kerker(a)saabgroup.com w. http://www.saabgroup.com

3 2

[Users] One way audio problem (Urgent)
by kyneoh＠dallab.com 11 Apr '07

11 Apr '07

Hello all, Ive been having a problem i cant seem to figure out, and i can see from various Google and list searches that i havent been the only one having this, but havent been able to find a solution yet for more than 3 weeks. Ive a bit frustrate to solve current problem. Ive got 4 UAs here. User agent A & C & D are located under WLAN, and user agent B located in the same LAN as SIP server and STUN server. User agent A & C connected to a restricted NAT that used STUN server to discover its own public IP. User agent A & C are mobile phone. (OS : window mobile 5.0) User agent D is softphone install in laptop. (OS : window XP) When calls gets sent from Box A to Box C (remote) i get a problem with the one-way audio(incoming caller can hear my voice, but i cant hear them). While, calls from Box A to Box D was successful. Both calls are form the same structure, how come calls work on A to D ??? (D) | (A)-----[NAT]-----------[NAT]---------------[SIP Proxy]-----(B) | | (C) [STUN server] *** SIP server and STUN server have assigned different public IP adress. SIP server : OPENSER + RTP Proxy in OPENSUSE Linux. Box A, C : Ageet softphone. Box B ,D : X-Lite softphone. Hope some of you out there are able to help me out a bit, its getting a bit annoying not being able to hear people calling me. Regards, Kum

2 1

[Users] Proxy server only
by Lucas Diaz 11 Apr '07

11 Apr '07

This is my scenario: I got Sip devices using as Registrar and Proxy a public service from Internet. I want to OpenSER in the middle so I can track and manage the calls. How can I configure OpenSER? Do I need to do something in the external Server? Any help would be apreciated. Lucas.

3 2

[Users] memlog question
by Papadopoulos Georgios 11 Apr '07

11 Apr '07

Hello all, I have a question about what is the best way to enable logging in openser 1.1.x. I want also to be able to get the memory dump so I have compiled with the appropriate flags and in my config I have debug=2 memlog=1 log_stderror=no This outputs everything into /var/log/messages, but the problem is that the memory dump section is messed up (lines do not seem to be in the right order). Also it takes a lot of time to stop openser with this configuration and I end up killing it. Is this because dumping all the memory to syslog is very slow? Is there a better way to do this? 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.

2 1

[Users] users in the same OpenSer cant call each other
by Joao Pereira 11 Apr '07

11 Apr '07

Hello My Openser users can call PSTN numbers through a PSTN gateway. But they cant call each other dialing the usernames. Where is the bug in my config? here is my openser.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # -------------------------------------------------- # Registration # -------------------------------------------------- if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.com", "subscriber")) { www_challenge("myserver.com", "0"); exit; }; save("location"); exit; }; } if (is_method("INVITE")) { # Route E.164 numbers to PSTN Gateway if (uri=~"sip:\+[1-9][0-9]*@myserver.com" ) { if (isflagset(29)){ route(2); return; } } # Route E.164 numbers to PSTN Gateway if (uri=~"sip:[1-9][0-9]*@myserver.com" ) { if (isflagset(29)){ route(2); return; } } } # method = invite 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; } # Gateway PSTN route[2] { rewritehost("gateways_IP"); t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } Thanks for the help Joao Pereira

2 1

[Users] help plz
by shruu thii 11 Apr '07

11 Apr '07

hi all, i'm using fc4.i've installed openser 1.0.1.how should i enable this for ipv6. i'm using linphone client.how do i register? i wanna do pc-pc calling on lan. thankyou. shruthi --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

2 1

[Users] force_rtp_proxy() and t_write_unix()
by Victor Gamov 11 Apr '07

11 Apr '07

hi! Can anybody explain me: when I do force_rtp_proxy("c") and then t_write_unix() which request body write to socket -- original or rewrited by force_rtp_proxy() ? Now I trying to configure OpenSER-1.2.0 to work with SEMS and I found that SEMS get original SDP with internal IPs. Is it my OpenSER misconfiguration? Thanks! -- CU, Victor Gamov

2 3

Re: [Serusers] re: Malformed CONTACT field in 302 response
by Olaf Bergmann 11 Apr '07

11 Apr '07

Jignesh Gandhi wrote: > Thank you for your reply. > > Options #1 and 2 are not possible. > I am looking at how to do Option 3. Can you tell me as to > how to go about setting the q value ? Sorry, I am not sure how to do this in SER 0.8.x. Maybe someone else out there? > thanks, > Jignesh gandhi > > > On 4/10/07, *Olaf Bergmann* <Olaf.Bergmann(a)freenet-ag.de > <mailto:Olaf.Bergmann@freenet-ag.de>> wrote: > > Jignesh Gandhi wrote: > > > I recently posted almost the same question. > > And you got a number of good suggestions how to get around your > problem. Largely, the options were: > > - Make your softswitch vendor fix the bug. > - Use another softswitch. > - Adapt your custom function gl_redirect() to create a Contact field > with angle brackets. > - Set the q value. > > Which one did you check out? Did it work? Why not? What about the > other options? > > Regards, > Olaf > > > > > -- > Jignesh Gandhi > jigpgandhi(a)gmail.com <mailto:jigpgandhi@gmail.com> Regards, Olaf

1 0

[Users] One way audio problem (Urgent)
by kyneoh＠dallab.com 11 Apr '07

11 Apr '07

1 0

[Users] Terminate Call
by Javier 11 Apr '07

11 Apr '07

How to I can teminate call when prepayed debit run out? __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

3 2

[Users] "detached" timer
by T.R. Missner 10 Apr '07

10 Apr '07

Does anyone know what causes this? */set_timer for 1 list called on a "detached" timer -- ignoring /* I also see */set_timer for 3 list called on a "detached" timer -- ignoring /* When this happens Openser seems to lock up for 10 seconds or so. From searching it appears this is caused by a race but I am not sure what the race is or why this results in an unresponsive openser instance for multiple seconds. Transaction expiration racing reply? Desperately need to understand how this could be triggered so I can get customer to adjust system. Any way to adjust? tried tweaking fr_inv_timer but no joy. TR

6 32

[Serusers] problem with compiling the new CVS release
by İlker Aktuna (Koç.net) 10 Apr '07

10 Apr '07

Hi, I have a working SER installation which includes presence modules etc. Today I tried to compile the latest CVS version because of a requirement to the reigstrar module. (NAT support with mediaproxy) unfortunately, the "make" process did not comlete. I got these errors: make[1]: Entering directory `/root/yeni/sip_router/modules/acc_db' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev2"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.0"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/root/ser2/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -c acc_db.c -o acc_db.o acc_db.c:164: error: expected â€˜=â€™, â€˜,â€™, â€˜;â€™, â€˜asmâ€™ or â€˜__attribute__â€™ before â€˜acc_dbfâ€™ acc_db.c:172: error: expected â€˜=â€™, â€˜,â€™, â€˜;â€™, â€˜asmâ€™ or â€˜__attribute__â€™ before â€˜keysâ€™ acc_db.c:173: error: expected â€˜=â€™, â€˜,â€™, â€˜;â€™, â€˜asmâ€™ or â€˜__attribute__â€™ before â€˜valsâ€™ acc_db.c: In function â€˜init_dataâ€™: acc_db.c:406: error: â€˜keysâ€™ undeclared (first use in this function) acc_db.c:406: error: (Each undeclared identifier is reported only once acc_db.c:406: error: for each function it appears in.) acc_db.c:407: error: â€˜valsâ€™ undeclared (first use in this function) acc_db.c: In function â€˜fmt2strarâ€™: acc_db.c:594: error: â€˜valsâ€™ undeclared (first use in this function) acc_db.c: In function â€˜log_requestâ€™: acc_db.c:755: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c:760: error: â€˜keysâ€™ undeclared (first use in this function) acc_db.c:760: error: â€˜valsâ€™ undeclared (first use in this function) acc_db.c: In function â€˜child_initâ€™: acc_db.c:974: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c: In function â€˜mod_destroyâ€™: acc_db.c:989: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c: In function â€˜mod_initâ€™: acc_db.c:1014: warning: implicit declaration of function â€˜bind_dbmodâ€™ acc_db.c:1014: error: â€˜acc_dbfâ€™ undeclared (first use in this function) acc_db.c:1020: warning: implicit declaration of function â€˜DB_CAPABILITYâ€™ acc_db.c:1020: error: â€˜DB_CAP_INSERTâ€™ undeclared (first use in this function) make[1]: *** [acc_db.o] Error 1 make[1]: Leaving directory `/root/yeni/sip_router/modules/acc_db' make: *** [modules] Error 1 # Does anyone have any idea what the problem is ? Thanks, ilker

8 25

[Users] help
by shruu thii 10 Apr '07

10 Apr '07

hi list, i'm a newbie to openser..i've installed openser,please do help me in proceeding further... --------------------------------- Check out what you're missing if you're not on Yahoo! Messenger

1 0

[Users] Segmentation fault when sip_trace is set to on-
by Sergio Gutierrez 09 Apr '07

09 Apr '07

Hi list. I am using openser 1.2.0, compiled to 64 bits on Solaris 10 on Sparc Platform. The service starts ok, but, when I send the sip_trace comand via fifo, the thread which writes on database crashes with segmentation fault. This is the backtrace of the core file: #0 0xffffffff7e990dc8 in strftime () from /lib/64/libc.so.1 #1 0xffffffff7be06a34 in time2mysql (_time=5051504780619285336, _result=0xffffffff7bf09f4c "", _res_len=-1) at /home/operador/openser-1.2.0-notls/modules/mysql//utils.c:54 #2 0xffffffff7be07724 in val2str (_c=0x900, _v=0, _s=0xffffffff7bf09f4c "", _len=0x2ff) at /home/operador/openser-1.2.0-notls/modules/mysql//val.c:147 #3 0xffffffff7be02480 in db_mysql_print_values (_c=0xffffff7fffebce00, _b=0xff <Address 0xff out of bounds>, _l=637645056, _v=0x32393038, _n=-1360931840) at /home/operador/openser-1.2.0-notls/modules/mysql//dbase.c:145 #4 0xffffffff7be041e4 in db_mysql_insert (_h=0xff, _k=0x10032393130, _v=0xffffff7ffff74000, _n=744703) at /home/operador/openser-1.2.0-notls/modules/mysql//dbase.c:584 #5 0xffffffff7910a5a8 in sip_trace (msg=0x101c0000, s1=0x2000 <Address 0x2000 out of bounds>, s2=0x2340 <Address 0x2340 out of bounds>) at /home/operador/openser-1.2.0-notls/modules/siptrace//siptrace.c:472 #6 0x000000010000d2a4 in do_action (a=0x0, msg=0x1000eb00046) at /home/operador/openser-1.2.0-notls//action.c:883 #7 0x000000010000e394 in run_top_route (a=0x10026fb78, msg=0x10028a0a0) at /home/operador/openser-1.2.0-notls//action.c:131 #8 0x000000010006503c in receive_msg (buf=0x100260 <Address 0x100260 out of bounds>, len=586177280, rcv_info=0xffffffff7ffff7a8) at /home/operador/openser-1.2.0-notls//receive.c:156 #9 0x00000001000a1900 in udp_rcv_loop () at /home/operador/openser- 1.2.0-notls//udp_server.c:465 #10 0x000000010003465c in main (argc=117, argv=0xffffff7910acf0ff) at /home/operador/openser-1.2.0-notls//main.c:1399 Thanks in advance for your help and attention. Kind regards, Sergio Gutiérrez EPM Telecomunicaciones.

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[Users] XCAP Lite 0.8 released
by Adrian Georgescu 09 Apr '07

09 Apr '07

Change log xcap-lite (0.8) unstable; urgency=low * Synchronize watchers table with presrules (requires domxml in php4) * Add database locking for PUT * Add more debugging information * Fix realm detection in hostname * Validate documents against standardized schemas (php 5 only) * Update documentation The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.8.tar.gz Regards, Adrian

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[Serusers] install SER
by Tuan Nguyen 09 Apr '07

09 Apr '07

Hello, I want to install SER in ubuntu. I've downloaded from cvs the version -D 06042007. I 've compiled and haven't no error, now i use "make install" to install but it didn't work without error or inform (I've installed build tools). Anyone can tell me what happen and how can i fix this problem? Thanks in advance T.

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[Users] avp_db_store can't write into database
by chungyu 09 Apr '07

09 Apr '07

Hi all, I am trying to use avp_db_store() stores my table "contact" with caller uri ex: table name: contact ----------------------------- | username | domain | | ----------------------------- | | | | ----------------------------- | | | | ----------------------------- If (method=="INVITE"){ avp_write("$ruri/username", "s:name"); avp_write("$ruri/domain", "s:domain"); avo_db_store("s:name", "s:username/contact") avp_db_store("s:domain", "s:domain/contact"); }; if the call bob then save username and domain of callee in contact callee: bob at iptel.org => ----------------------------- | username | domain | | ----------------------------- | bob | iptel.org | | ----------------------------- | | | | ----------------------------- but not success, system reply ERROR:avpops:db_store: insert faild Am I setting wrong param for avo_db_store()? Thanks a lot in advance chungyu

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[Users] o_uri in 1.2
by Tim 06 Apr '07

06 Apr '07

Does anyone know how I can get hold of the o_uri ( outgoing uri ) for accounting in openser 1.2? I thought it might be destination uri but that didn't work. Thanks Tim

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[Users] cancels not going through
by Vadim Berezniker 06 Apr '07

06 Apr '07

A number is dialed on a phone and the INVITE goes: PHONE => ASTERISK => SER => PHONE If the other phone doesn't pickup and the calling phone is hung up, it sends a BYE to Asterisk. Asterisk sends a CANCEL to SER. SER cancels its transaction but it doesn't relay the CANCEL to the phone so it keeps ringing. Am I missing something here?

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Re: [Users] proxy-authenticate causes phone to drop authenticate creds
by Daniel-Constantin Mierla 06 Apr '07

06 Apr '07

Try to run asterisk in ddebug mode, and watch debug messages, maybe you get some hints why it considers as coming from phone and not from openser. Cheers, Daniel On 04/05/07 18:00, Mark Price wrote: > Daniel, > I tried that. > I think that asterisk still sees the request as coming from the phone > instead of from openser. Is there possibly something I'm doing wrong? > Is there something I probably need to do extra? > > Here is my openser.cfg: >

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Re: [Users] Presence: nothing to retr ansmit ???
by Daniel-Constantin Mierla 06 Apr '07

06 Apr '07

Hello, can you capture and send the network trace for such case (using ngrep or wireshark)? Cheers, Daniel On 04/06/07 04:28, Eddie wrote: > Hi Daniel, > Thank you very much for your advise. > I captured packets by ethereal, but there is no reply for the first > PUBLISH message. > To implement PUBLISH module, I just followed the README in > /openser..../modules/presence > Is there anything else shoule be considerd? > Thanks so much! > Brs, > Eddie > > 在2007-04-05，"Daniel-Constantin Mierla" <daniel(a)voice-system.ro> 写道： > > Hello, > > seems that this is a retransmission of PUBLISH. Check the log above the > pasted messages. There should be the messages of processing the initial > PUBLISH. Check there to see if the reply was sent (you can use > ngrep/wireshark to watch the SIP traffic, therefore, you can see wheter > reply is sent or not). > > Cheers, > Daniel > > > On 04/03/07 10:58, Eddie wrote: > > Hi all, > > I am a new to openser. I am using openser-1.2.0-notls. I met a issue > > when I setup a presence server. > > I use a UDP testing tool to send PUBLISH message : > > PUBLISH sip:111@sip.comverse.com <mailto:111@sip.comverse.com> SIP/2.0 > > To: <sip:111@sip.comverse.com <mailto:111@sip.comverse.com>> > > From: <sip:111@sip.comverse.com <mailto:111@sip.comverse.com>>;tag=aa0e9e60 > > Via: SIP/2.0/UDP > > 172.19.1.190:5060;branch=z9hG4bK-d87543-4f7005092661b51b-1--d87543-;rport > > Call-ID: dd2288324976454c@UG9ja2V0X1BD > > <mailto:dd2288324976454c@UG9ja2V0X1BD> > > CSeq: 4444 PUBLISH > > Contact: <sip:111@172.19.1.190 <mailto:111@172.19.1.190>:5060> > > Expires: 3600 > > Max-Forwards: 70 > > Content-Type: application/pidf+xml > > Event: presence > > Content-Length: 549 > > <?xml version="1.0" encoding="UTF-8" ?> > > <presence xmlns="urn:ietf:params:xml:ns:pidf" > > xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" > > xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" > > xmlns:op="urn:oma:params:xml:ns:pidf:oma-pres" > > xmlns:xsi="http://www.w3.org/2001/XMLSchema-instan > > <tuple id="t0001"> > > <status> > > <basic>open</basic> > > </status> > > <op:overriding-willingness> > > <op:basic>open</op:basic> > > </op:overriding-willingness> > > </tuple> > > </presence> > > but no 200 OK return back from the server. and I set debug level to 7, > > the log is as followed: > > 4(5485) method: <PUBLISH> > > 4(5485) uri: <sip:111@sip.comverse.com <mailto:111@sip.comverse.com>> > > 4(5485) version: <SIP/2.0> > > 4(5485) parse_headers: flags=2 > > 4(5485) DEBUG:parse_to:end of header reached, state=10 > > 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com <mailto:111@sip.comverse.com>} > > 4(5485) DEBUG: get_hdr_field: <To> [28]; uri=[sip:111@sip.comverse.com <mailto:111@sip.comverse.com>] > > 4(5485) DEBUG: to body [<sip:111@sip.comverse.com <mailto:111@sip.comverse.com>> > > > > 4(5485) Found param type 232, <branch> = > > <z9hG4bK-d87543-4f7005092661b51b-1--d87543->; state=6 > > 4(5485) Found param type 235, <rport> = <n/a>; state=17 > > 4(5485) end of header reached, state=5 > > 4(5485) parse_headers: Via found, flags=2 > > 4(5485) parse_headers: this is the first via > > 4(5485) After parse_msg... > > 4(5485) preparing to run routing scripts... > > 4(5485) parse_headers: flags=100 > > 4(5485) get_hdr_field: cseq <CSeq>: <4445> <PUBLISH> > > 4(5485) DEBUG:maxfwd:is_maxfwd_present: value = 70 > > 4(5485) DEBUG: add_param: tag=aa0e9e60 > > 4(5485) DEBUG:parse_to:end of header reached, state=29 > > 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com <mailto:111@sip.comverse.com>} > > 4(5485) parse_headers: flags=200 > > 4(5485) DEBUG: get_hdr_body : content_length=549 > > 4(5485) found end of header > > 4(5485) find_first_route: No Route headers found > > 4(5485) loose_route: There is no Route HF > > 4(5485) grep_sock_info - checking if host==us: 16==12 && > > [sip.comverse.com] == [172.19.1.170] > > 4(5485) grep_sock_info - checking if port 5060 matches port 5060 > > 4(5485) grep_sock_info - checking if host==us: 16==12 && > > [sip.comverse.com] == [172.19.1.170] > > 4(5485) grep_sock_info - checking if port 5060 matches port 5060 > > 4(5485) DEBUG: t_newtran: T on entrance=0xffffffff > > 4(5485) parse_headers: flags=ffffffffffffffff > > 4(5485) parse_headers: flags=78 > > 4(5485) t_lookup_request: start searching: hash=29736, isACK=0 > > 4(5485) DEBUG: RFC3261 transaction matched, > > tid=-d87543-4f7005092661b51b-1--d87543- > > 4(5485) DEBUG:tm:REF_UNSAFE: after is 1 > > 4(5485) DEBUG: t_lookup_request: transaction found (T=0xb6148f48) > > 4(5485) DEBUG:tm:t_retransmit_reply: nothing to retransmit > > 4(5485) DEBUG:tm:UNREF_UNSAFE: after is 0 > > 4(5485) DEBUG:destroy_avp_list: destroying list (nil) > > 4(5485) receive_msg: cleaning up > > 5(5488) PRESENCE: msg_watchers_clean:cleaning pending subscriptions > > 5(5488) PRESENCE: msg_active_watchers_clean:cleaning expired watcher > > information > > 5(5488) PRESENCE:msg_presentity_clean:cleaning expired presentity > > information > > I am confused about the issue, and I am not sure whether it is caused > > by configuration files or mysql database. > > Please help me. Thanks so much ! > > Brs, > > Eddie > > > > > > ------------------------------------------------------------------------ > > 每天上班第一件要做的事情是？马上看看>> <http://www.126.com/> > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ------------------------------------------------------------------------ > 每天上班第一件要做的事情是？马上看看>> <http://www.126.com/>

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