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[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

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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.

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[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

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[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.

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[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

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[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-)

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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

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[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.

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[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

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[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

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[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

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[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.

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[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

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[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.

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[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

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[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-----

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[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

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[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

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[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

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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.

2 1

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…>

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[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

4 5

[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.

1 0

[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

1 0

[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.

1 0

[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

2 4

[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

2 3

[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?

2 3

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: >

1 0

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/>

1 0

[Users] Help wanted
by Flavio Goncalves 06 Apr '07

06 Apr '07

Dear coleagues, I need help to translate an OpenSER eBook (building a SIP provider) from portuguese to english. I'm willing paying US$75,00 (paypal) per chapter. If any of you have interest, please contact me at flavio(a)asteriskguide.com Regards, Flavio E. Goncalves

1 0

[Serusers] Can SER do it? Extract info
by Gustavo Passos Tourinho 05 Apr '07

05 Apr '07

Hi There. Can SER extract info from request uri? For example, If I have this uri: INVITE sip:123456#0159559432@my.sip.domain How can I get data before "#" (123456)? I dont want to "strip" this information, just want to get it for further utilization. Is that possible? If it can be done, can be done using variable number of digits? For example: INVITE sip:123#0159559432@my.sip.domain INVITE sip:1234567#0159559432@my.sip.domain Thanks in advanced. Regards, Gustavo

2 1

[Serusers] support to implemente multi leg call radius accounting
by Rosario Pingaro 05 Apr '07

05 Apr '07

We are looking to find support to implement multi leg call accounting in our 0.9.6 ser configuration. We use radius accounting. Any consultant interested supporting us may submit a quotation to this email. Reagrds Rosario

1 0

[Users] t_relay() has no effect !
by JP Ramoul 05 Apr '07

05 Apr '07

2 1

[Serusers] uninstall
by Tuan Nguyen 05 Apr '07

05 Apr '07

Hi everybody, I have installed ser-0.9.6 and now i want to uninstall it to new install in correctly way because it works not property. how can i complete uninstall old version. thanks in advance T.

2 1

Re: [Serusers] error by ser-2.1
by Ladislav Andel 05 Apr '07

05 Apr '07

Here is the answer: http://lists.iptel.org/pipermail/serusers/2007-April/032657.html or http://lists.iptel.org/pipermail/serusers/2007-April/032655.html You need to download stable or earlier version of SER 2.1 because of temporary broken database API. Lada Tuan Nguyen wrote: > Hi Andel, > > I do not see anywhere say that "2 posts before question". > > I am newbie by SER, I have many problem to try to run it, how can > solve this problem without question? and what is "Support for users of > the SIP Express Router" in http://lists.iptel.org/mailman/listinfo/ ? > > Please tell me what should i do to solve my problem when i don`t find > answer in archive? > thanks, > T. > On 4/5/07, *Ladislav Andel* <ladaan(a)iptel.org > <mailto:ladaan@iptel.org>> wrote: > > Hi, > Have you been reading the serusers mailing list ? > Just 2 posts before your question you will find the answer. > > It is for everybody. Read and search the list before asking. > > Peace :), > Lada > > >

1 0

[Serusers] error by ser-2.1
by Tuan Nguyen 05 Apr '07

05 Apr '07

hi everyone, when i compile new version i get follow error: make[1]: Entering directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' make[1]: Leaving directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' make[1]: Entering directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_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+cvs20070404"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -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 authdb_mod.c -o authdb_mod.o In file included from authdb_mod.c:49: authdb_mod.h:57: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'auth_dbf' authdb_mod.c:106: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'auth_dbf' authdb_mod.c: In function 'child_init': authdb_mod.c:165: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c:165: error: (Each undeclared identifier is reported only once authdb_mod.c:165: error: for each function it appears in.) authdb_mod.c: In function 'mod_init': authdb_mod.c:180: warning: implicit declaration of function 'bind_dbmod' authdb_mod.c:180: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c: In function 'destroy': authdb_mod.c:206: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c: In function 'authdb_fixup': authdb_mod.c:226: error: 'auth_dbf' undeclared (first use in this function) authdb_mod.c:231: warning: implicit declaration of function 'table_version' make[1]: *** [authdb_mod.o] Error 1 make[1]: Leaving directory `/net/u/ntm/Desktop/ser- 2.1.0+cvs20070404/modules/auth_db' make: *** [modules] Error 1 anyone know why and how correct it. thanks T.

2 1

[Users] Advanced OpenSER Programming Course, Paris, INRIA, June 4-6, 2007
by Daniel-Constantin Mierla 05 Apr '07

05 Apr '07

Hello, following some discussions started during first OpenSER Summit, we are happy to announce that first Advanced Programming Course will happen very soon. Courtesy of INRIA (many thanks to Philippe Sultan) we have the room and a good connected location. There will be three full days of deep analysis and survey of internal architecture and structures, best practices to add new features, optimizations and OpenSER's specific components (memory manager, locking system, ...). The course is dedicated to people willing to develop OpenSER extensions. It will be a class of 14 persons. The participation fee will be just to cover the logistic required for this event, estimated to be about 150-200Euro per person, for the full course (mainly is subject of how many participants will be registered). Teachers will be Bogdan-Andrei Iancu and Daniel-Constantin Mierla, from OpenSER development team, and the primary goal is to attract more developers to core components of OpenSER and build a higher development squad. Also, there will be open discussions to improve existing code and functionalities. If the interest for this event is high, the primary selection criteria will include the level of contributions so far and willingness to contribute in the future. New similar events will be scheduled if the number of people showing interest is enough for another one or more classes. For this one, we expect to have people with pretty good knowledge of SIP and OpenSER operation, to avoid long introduction lessons. If you are interested in this course, please send an email to team(a)openser.org. <mailto:team@openser.org.> News link: http://www.openser.org/mos/view/News/NewsItem/Advanced-OpenSER-Programming-… Cheers, Daniel

1 1

[Users] Presence: nothing to retransmit ???
by Eddie 05 Apr '07

05 Apr '07

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 SIP/2.0 To: <sip:111@sip.comverse.com> From: <sip: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 CSeq: 4444 PUBLISH Contact: <sip: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="<A href="http://www.w3.org/2001/XMLSchema-instan">http://www.w3.org/2001/XMLSchema-instan</A><BR> <tuple id="t0001"><BR> <status><BR> <basic>open</basic><BR> </status><BR> <op:overriding-willingness><BR> <op:basic>open</op:basic><BR> </op:overriding-willingness><BR> </tuple><BR></presence></DIV> <DIV> </DIV> <DIV>but no 200 OK return back from the server. and I set debug level to 7, the log is as followed: </DIV> <DIV> </DIV> <DIV>4(5485) method: <PUBLISH><BR> 4(5485) uri: <sip:111@sip.comverse.com><BR> 4(5485) version: <SIP/2.0><BR> 4(5485) parse_headers: flags=2<BR> 4(5485) DEBUG:parse_to:end of header reached, state=10<BR> 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com}<BR> 4(5485) DEBUG: get_hdr_field: <To> [28]; uri=[sip:111@sip.comverse.com]<BR> 4(5485) DEBUG: to body [<sip:111@sip.comverse.com><BR><BR> 4(5485) Found param type 232, <branch> = <z9hG4bK-d87543-4f7005092661b51b-1--d87543->; state=6<BR> 4(5485) Found param type 235, <rport> = <n/a>; state=17<BR> 4(5485) end of header reached, state=5<BR> 4(5485) parse_headers: Via found, flags=2<BR> 4(5485) parse_headers: this is the first via<BR> 4(5485) After parse_msg...<BR> 4(5485) preparing to run routing scripts...<BR> 4(5485) parse_headers: flags=100<BR> 4(5485) get_hdr_field: cseq <CSeq>: <4445> <PUBLISH><BR> 4(5485) DEBUG:maxfwd:is_maxfwd_present: value = 70<BR> 4(5485) DEBUG: add_param: tag=aa0e9e60<BR> 4(5485) DEBUG:parse_to:end of header reached, state=29<BR> 4(5485) DBUG:parse_to: display={}, ruri={sip:111@sip.comverse.com}<BR> 4(5485) parse_headers: flags=200<BR> 4(5485) DEBUG: get_hdr_body : content_length=549<BR> 4(5485) found end of header<BR> 4(5485) find_first_route: No Route headers found<BR> 4(5485) loose_route: There is no Route HF<BR> 4(5485) grep_sock_info - checking if host==us: 16==12 && [sip.comverse.com] == [172.19.1.170]<BR> 4(5485) grep_sock_info - checking if port 5060 matches port 5060<BR> 4(5485) grep_sock_info - checking if host==us: 16==12 && [sip.comverse.com] == [172.19.1.170]<BR> 4(5485) grep_sock_info - checking if port 5060 matches port 5060<BR> 4(5485) DEBUG: t_newtran: T on entrance=0xffffffff<BR> 4(5485) parse_headers: flags=ffffffffffffffff<BR> 4(5485) parse_headers: flags=78<BR> 4(5485) t_lookup_request: start searching: hash=29736, isACK=0<BR> 4(5485) DEBUG: RFC3261 transaction matched, tid=-d87543-4f7005092661b51b-1--d87543-<BR> 4(5485) DEBUG:tm:REF_UNSAFE: after is 1<BR> 4(5485) DEBUG: t_lookup_request: transaction found (T=0xb6148f48)<BR> 4(5485) DEBUG:tm:t_retransmit_reply: nothing to retransmit<BR> 4(5485) DEBUG:tm:UNREF_UNSAFE: after is 0<BR> 4(5485) DEBUG:destroy_avp_list: destroying list (nil)<BR> 4(5485) receive_msg: cleaning up<BR> 5(5488) PRESENCE: msg_watchers_clean:cleaning pending subscriptions<BR> 5(5488) PRESENCE: msg_active_watchers_clean:cleaning expired watcher information<BR> 5(5488) PRESENCE:msg_presentity_clean:cleaning expired presentity information</DIV> <DIV> </DIV> <DIV>I am confused about the issue, and I am not sure whether it is caused by configuration files or mysql database.</DIV> <DIV> </DIV> <DIV>Please help me. Thanks so much !</DIV> <DIV> </DIV> <DIV> </DIV> <DIV>Brs,</DIV> <DIV> </DIV> <DIV>Eddie</DIV> <DIV> </DIV>

2 1

[Users] host identification
by Joao Pereira 05 Apr '07

05 Apr '07

Hello does openSER uses the host identificayion in any way to authenticate its costumers? Because I re-installed my server and thats the only difference from the previous instalation (its even in the same machine). Thanks regards Joao pereira

2 1

[Users] openser 1.2 avp
by laurent schweizer 05 Apr '07

05 Apr '07

Hello, with openser 1.0 I have this if ( avp_db_load("$low_c_service/username","i:/serviceLowCoastredirect")) and for openser 1.2 I try to do this if ( avp_db_load("$avp(low_c_service)/username","i:/serviceLowCoastredirect")) or if ( avp_db_load("$avp($low_c_service)/username","i:/serviceLowCoastredirect")) low_c_service is an alias avp_aliases="low_c_service=i:500" but I have always an error. 0(0) ERROR:avops:parse_avp_db:: error - bad avp flags 0(0) ERROR:avpops:fixup_db_avp: parse failed 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 633 0(0) ERROR: fix_expr : fix_actions error

2 1

[Users] SIP SUBSCRIBE AND MESSAGE msgs
by Abdurrahman Sahin 05 Apr '07

05 Apr '07

hi all; i installed openser now trying to setup a basic text commnuication between two clients on the same pc (one client is windows msngr and the other one is x-lite ) it seems to me SUBSCRIBE and MESSAGE sip msgs are not processed. i generated openser.cfg with the wizard at sipwise.com site here is my ngrep dump for SUBSCRIBE msg ==================== U ua_public_ip:29350 -> 192.168.200.2:5060 SUBSCRIBE sip:burak@testsrv200 SIP/2.0. Via: SIP/2.0/UDP ua_public_ip:29347. Max-Forwards: 70. From: "apo@testsrv200" <sip:apo@testsrv200>;tag=29774bb79657418eb518068a22304b7c;epid=13d631ff91. To: <sip:burak@testsrv200>. Call-ID: 15088cb5e87d4fb887cc073d0fd9df07. CSeq: 1 SUBSCRIBE. Contact: <sip:ua_public_ip:29347>. User-Agent: RTC/1.3. Event: presence. Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml. Supported: com.microsoft.autoextend. Supported: ms-benotify. Proxy-Require: ms-benotify. Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip:29350 SIP/2.0 501 Method Not Supported Here. Via: SIP/2.0/UDP ua_public_ip:29347;rport=29350. From: "apo@testsrv200" <sip:apo@testsrv200>;tag=29774bb79657418eb518068a22304b7c;epid=13d631ff91. To: <sip:burak@testsrv200>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea73. Call-ID: 15088cb5e87d4fb887cc073d0fd9df07. CSeq: 1 SUBSCRIBE. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. MESSAGE msg (while trying to establish a text chat) =========================== U ua_public_ip:29355 -> 192.168.200.2:5060 MESSAGE sip:apo@testsrv200 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.33:58348;branch=z9hG4bK-d87543-2a3b75359734fe48-1--d87543-;rport. Max-Forwards: 70. To: "apo"<sip:apo@testsrv200>. From: "burak"<sip:burak@testsrv200>;tag=fb1a9f68. Call-ID: NTNhMGZlMmYwODIxNmFlNmNmYThmYjQ0NzFhOGFjODI.. CSeq: 2 MESSAGE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/im-iscomposing+xml. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 259. . <?xml version='1.0' encoding='UTF-8'?> <isComposing xmlns='urn:ietf:params:xml:ns:im-iscomposing' xmlns:xsi='http://www.w3.org/2001/XMLSchema-instance'> <state>active</state> <contenttype>goober</contenttype> <refresh>60</refresh> </isComposing> # U 192.168.200.2:5060 -> ua_public_ip:29355 SIP/2.0 501 Method Not Supported Here. Via: SIP/2.0/UDP 192.168.1.33:58348;branch=z9hG4bK-d87543-2a3b75359734fe48-1--d87543-;rport=29355;received=ua_public_ip. To: "apo"<sip:apo@testsrv200>;tag=329cfeaa6ded039da25ff8cbb8668bd2.402d. From: "burak"<sip:burak@testsrv200>;tag=fb1a9f68. Call-ID: NTNhMGZlMmYwODIxNmFlNmNmYThmYjQ0NzFhOGFjODI.. CSeq: 2 MESSAGE. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. . .

2 1

[Users] Strange behaviour on siptrace module
by Sergio Gutierrez 05 Apr '07

05 Apr '07

Hi list. I am having a problem with siptrace module. When I send the command sip_trace on through fifo, openser do not send back responses to user agents. On logfile (with debug=9), I just receive this message: sip_trace: storing info... On database do not appear any trace message. Thanks in advance for your attention. Kind regards. Sergio G.

2 1

[Users] proxy-authenticate causes phone to drop authenticate creds
by Mark Price 05 Apr '07

05 Apr '07

Hi, I am using openser 1.1.x as a front end to asterisk. Here is what happens in the dialog: 1. phone sends invite 2. openser says "Unauthorized" 3. phone sends invite + Authorization header 4. openser auths the packet and forwards to asterisk 5. asterisk says "proxy authentication required" 6. openser says "proxy authentication required" here is where it gets interesting. 7.a On some devices, the phone sends invite + Authorization header + Proxy-Authorizaiton header. 7.b On some devices, the phone sends invite + Proxy-Authorization header. How does one work around 7.b? Really, what I want to see is how to make it so that asterisk does not try to authenticate the call. However, since that doesn't seem to be possible, what is the sensible way to route these calls? Thanks, Mark Price

2 1

[Users] Authentication problem
by charl van vuuren 05 Apr '07

05 Apr '07

I'm trying to setup Openser (1.1) as registrar with mediaproxy and mysql all on the same box. I am currently using the SIP Proxy/Registrar with Mediaproxy config from sipwise. MySQL and Mediaproxy are running. Xlite registers successfully with Openser with the users I have created. The problem comes with the authentication. Both www_authorize and proxy_authorize fail. www_authorize fails at regtistration and proxy_authorize off course then fails when I try make a call. Any suggestions? Thanx

2 1

[Users] user agent in avp
by Mark Price 05 Apr '07

05 Apr '07

Hi, In openser 1.1.x, is it possible to get similar functionality to that suggested by the following: avp_check($hdr(ua[0]),"re/Some Phone UA/") thanks Mark Price

2 1

[Users] radius attributes
by Terry Robert 05 Apr '07

05 Apr '07

Hi openser, Following the tutorial on FreeRadius integration (http://www.openser.org/docs/openser-radius-1.0.x.html), I'm seeing accounting packets successfully sent to Radiusd. However, some attributes are missing, notably: UserName, CallingStationId, and CalledStationId. Radius client dictionary contains: ATTRIBUTE User-Name 1 string ATTRIBUTE Password 2 string ... ATTRIBUTE Called-Station-Id 30 string ATTRIBUTE Calling-Station-Id 31 string Output from 'radiusd -X' shows: ... Module: Instantiated detail (detail) Listening on authentication *:1812 Listening on accounting *:1813 Ready to process requests. rad_recv: Accounting-Request packet from host 10.10.90.198:32812, id=10, length=120 Acct-Status-Type = Start Service-Type = IAPP-Register Attr-102 = 0x000000c8 Error-Cause = 1 Event-Timestamp = "Apr 4 2007 15:10:27 PDT" Attr-105 = 0x3937333153495070546167303031 Attr-104 = 0x3937323853495070546167303131 Acct-Session-Id = "1-9731(a)10.10.90.51" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 10.10.90.198 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 0 modcall[preacct]: module "preprocess" returns noop for request 0 rlm_acct_unique: WARNING: Attribute User-Name was not found in request, unique ID MAY be inconsistent ... Any tips on how to fix this? Thanks.

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RE: [Serusers] Problems with ACK
by Jiri Kuthan 04 Apr '07

04 Apr '07

I think the key message for problem analysis, client's ACK, is missing. This is the setting, right:? .100/UAC -----NAT/.248----> SER outbound @ .246 -----> UAS @ .239 The ACK in your message dump is only that one sent from proxy (246) to UAS (.239). The ACK which can possibly mismatch and produce the error message you mentioned must come from upstream. To be properly formatted, it must have Via identical to that of initial INVITE ( Via: SIP/2.0/UDP 192.168.1.100:5060;rport=46127;received=100.100.100.248;branch=z9hG4bK2893084087) -jiri At 17:50 27/03/2007, Ricardo Martinez wrote: No. Time Source Destination Protocol Info > 14 6.102719 100.100.100.239 100.100.100.246 SIP Status: 487 Request Terminated >Session Initiation Protocol > Status-Line: SIP/2.0 487 Request Terminated > Status-Code: 487 > Resent Packet: False > Message Header > Via: SIP/2.0/UDP 100.100.100.246;branch=z9hG4bK895c.a98d35.0 > Via: SIP/2.0/UDP 192.168.1.100:5060;rport=46127;received=100.100.100.248;branch=z9hG4bK2893084087 >No. Time Source Destination Protocol Info > 15 6.103011 100.100.100.246 100.100.100.239 SIP Request: ACK sip:005622408196@100.100.100.239 >Session Initiation Protocol > Request-Line: ACK sip:005622408196@100.100.100.239 SIP/2.0 > Method: ACK > Resent Packet: False > Message Header > Via: SIP/2.0/UDP 100.100.100.246;branch=z9hG4bK895c.a98d35.0 > f: "cl" <sip:cl@sipvoiss.desa.redvoiss.net>;tag=1587516403 > SIP Display info: "cl" > SIP from address: sip:cl@sipvoiss.desa.redvoiss.net > SIP tag: 1587516403 > i: <mailto:e94a98aa177e8579e5242a2091c0ac5f@192.168.1.100>e94a98aa177e8579e5242a2091c0ac5f(a)192.168.1.100 > To: <sip:0299005622408196@sipvoiss.desa.redvoiss.net>;tag=9146c297a4 >Thanks > >Ricardo.- >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

3 4

[Serusers] error with CPL script( CPL module of SER)
by dillip parida 04 Apr '07

04 Apr '07

Hi all, I am trying to implement call redirection feature using cpl script. I have complied cpl module that supports the cpl script. I have uploaded the cpl script. plz help me to solve the problem.. I have written this mail regarding the error that > we are getting in testing the cpl script of Call redirection. > > We have three X-lite phones with SIP uris 2001(a)172.16.1.24, > ,2002(a)172.16.1.28 and 2003(a)172.16.1.31 and all the three soft phones > have been registered with SER proxy running on 10.2.3.123. All the > three users have added in the SER database along with mail ids. > > user mail id > 2001 2001(a)172.16.1.24 > 2002 2002(a)172.16.1.28 > 2003 2003(a)172.16.1.31 > > The call scenario is as follows.. > 1.user 2001 will make a call to user 2002. > 2. If the user 2002 is either not responding (No answer) or busy, then > the call > will be redirected to user 2003. > 3. we able to make a call to 2002,and user 2002's X-lite phone is > ringing but the call is > not getting redirected. > > The cpl script for redirection has been deployed in SER proxy. > > I am sending the call log of sip phone:2001 as an attachment along > with snapshot sip phone 2001. > > I am sending cpl script at the bottam of this mail.plz find the cpl script at the end of this mail. > As shown in the xlite screen "500 CPL script execution failed" can u > please suggest me if there is > any error in XML/CPL syntax. > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------ > > <?xml version="1.0"?> > <!DOCTYPE html cpl PUBLIC "-//IETF//DTD RFCxxx CPL 1.0//EN" "cpl.dtd"> > <cpl> > <subaction id="pda"> > <location url="sip:2003@172.16.1.31 <sip:1003@172.16.1.31>"> > <proxy/> > </location> > </subaction> > <incoming> > <location url="sip:2002@172.16.1.28 <sip:1002@172.16.1.28>"> > <proxy timeout="8"> > <busy> > <sub ref="pda"/> > </busy> > <noanswer> > <sub ref="pda"/> > </noanswer> > </proxy> > </location> > </incoming> > </cpl>

1 0

Re: [Users] does openser support QoS ?
by NEILL WILKINSON 04 Apr '07

04 Apr '07

The more obvious question is: How does it know there is no more bandwidth available? As a signalling proxy, surely it has no visibility of the RTP usages or requirements. I guess if it counted all the complete INVITE-> OK dialogues and the SDP codec used it could estimate usage? But the RTP/media path could be different in each case of these so again the same question as above...... Neill...;o) ----- Original Message ---- From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)openser.org; yanlin <yanlin(a)fortinet.com> Sent: Tuesday, 3 April, 2007 10:41:31 AM Subject: Re: [Users] does openser support QoS ? On Tuesday 03 April 2007 11:15, yanlin wrote: > Hi, > > does openser support some kind of QoS ? > like, band width management, when there has no more usable bandwidth, then > new INVITE should be rejected. Hi Yanlin, if the bandwith is exceeded, no new INVITEs will get to the server and will timeout. :-) The "pike" module does some kind of DOS protection, but this is also no real QoS. There was a discussion two weeks ago about this topic on devel (the topic was "rand function in config file"), Klaus suggested some kind of bandwith management module there. But at the moment such a module don't exist in OpenSER. Cheers, Henning _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] New features added to OpenSer - BLA support and XMPP presence gateway
by Anca-Maria Vamanu 04 Apr '07

04 Apr '07

Two new modules were just introduced to SVN -- both are related to presence services. 1) pua_bla - implements Bridged Line Appearances according to the specifications in draft-anil-sipping-bla-03.txt. BLA, aka Shared Call/Line Appearance, comes from old telephony, where many devices share same line. Translated to SIP, means that many SIP devices share same AoR, but only one call can be established at a time, either addressed or initiated from the line. The call can be put on hold, and the others devices are notified so that they can pick up the call (the typical example is that with the secretary answering the call and putting it on hold to be picked up by the boss). The SIP devices must have client-side support for BLA as well. In out tests we used POLYCOM SoundPoint IP 430 SIP phones. The behaviour for this specific device: when picking up the handset form one phone a red light appears at the other phones. If attempting to call from a second phone using the same line, you get the message "Resources full". If a call is established, when putting it on hold, the light starts to flicker at the other phones and it is possible to pick it up. 2) pua_xmpp - is a presence gateway between SIMPLE and XMPP. It comes to complete the IM gateway, by enabling presence state information exchange between SIP/SIMPLE and Jabber/XMPP clients. For testings we used Gaim 2.0.0beta5, Psi v0.10 for XMPP and Xlite v3.0 for SIP. Some sample config file to implement above features will be published soon on dokuwiki.

1 0

RE: [Users] Problem with dialog module
by Leonid Fainshtein 04 Apr '07

04 Apr '07

Hi Bogdan, Please check the ngrep output. I put my comments along with the trace: 1. ****** The 203(a)172.17.4.22 calls 3525(a)172.17.4.22 U 2007/03/18 12:54:09.758887 172.17.2.11:5060 -> 172.17.4.22:5060 INVITE sip:3525@172.17.4.22:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5060 ;branch=z9hG4bK3796032784-846..Max-Forwards: 70..Allow: INVITE,OPTIONS,CANC EL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip :3525@172.17.4.22:5060>..Call-ID: 3795992784-846..CSeq: 2 INVITE..Contact: <sip:203@172.17.2.11:5060>..Content-Type: application/sdp..Proxy-Authorizat ion: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63 342cc574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b 864 a4ab937d17f42c7cd45a",algorithm=md5..Content-Length: 237....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 R TP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TELEPHONE-EVENT/8000.. a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 ver2..a=sendrecv.. U 2007/03/18 12:54:09.766562 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.17 .2.11:5060;branch=z9hG4bK3796032784-846..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>..Call-ID: 3795992784-84 6..CSeq: 2 INVITE..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Content-L ength: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4003 re q_src_ip=172.17.2.11 req_src_port=5060 in_uri=sip:3525@172.17.4.22:5060 out _uri=sip:3525@172.17.2.14;user=phone via_cnt==1".... U 2007/03/18 12:54:09.766871 172.17.4.22:5060 -> 172.17.2.14:5060 INVITE sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip:172.17.4 .22;lr=on;ftag=hssUA_3795992784-846;did=55f.4491d5b2>..Via: SIP/2.0/UDP 172 .17.4.22;branch=z9hG4bKbd8b.80bd0bc3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;b ranch=z9hG4bK3796032784-846..Max-Forwards: 69..Allow: INVITE,OPTIONS,CANCEL ,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS 3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3 525@172.17.4.22:5060>..Call-ID: 3795992784-846..CSeq: 2 INVITE..Contact: <s ip:203@172.17.2.11:5060>..Content-Type: application/sdp..Content-Length: 23 7....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t= 0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TE LEPHONE-EVENT/8000..a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 v er2..a=sendrecv.. U 2007/03/18 12:54:09.772591 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKbd8b.80bd0bc 3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784-846..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5 060>..Call-ID: 3795992784-846..CSeq: 2 INVITE..User-Agent: Grandstream BT11 0 1.0.8.16..Content-Length: 0.... U 2007/03/18 12:54:09.775043 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKbd8b.80bd0b c3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784-846..Recor d-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;did=55f.4491d5b2> ..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.1 7.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 INVITE. .User-Agent: Grandstream BT110 1.0.8.16..Content-Length: 0.... U 2007/03/18 12:54:09.777250 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK379603 2784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;di d=55f.4491d5b2>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846.. CSeq: 2 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Content-Length: 0.. .. U 2007/03/18 12:54:13.673700 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKbd8b.80bd0bc3.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;did=55f.4491d5b2>..Fro m: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.2 2:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 INVITE..User -Agent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=pho ne>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Type: application/sdp..Supported: replaces..Content-Length: 153....v=0. .o=3525 8000 8000 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 172.17.2.14..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU/8000..a=ptime:20.. U 2007/03/18 12:54:13.676298 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3796032784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846;did=55f .4491d5b2>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip: 3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.1 7.2.14:5060;user=phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS, INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: replaces..Content -Length: 153....v=0..o=3525 8000 8000 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 172.17.2.14..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU /8000..a=ptime:20.. U 2007/03/18 12:54:13.721732 172.17.2.11:5060 -> 172.17.4.22:5060 ACK sip:3525@172.17.2.14:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2 .11:5060;branch=z9hG4bK3799992784-846..Route: <sip:172.17.4.22;lr=on;ftag=h ssUA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Authorizatio n: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c6334 2cc574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b86 4a4 ab937d17f42c7cd45a",algorithm=md5..User-Agent: Claro BS3202 A454..From: <si p:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060 >;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 2 ACK..Contact: <sip :203@172.17.2.11:5060>..Content-Length: 0.... U 2007/03/18 12:54:13.724056 172.17.4.22:5060 -> 172.17.2.14:5060 ACK sip:3525@172.17.2.14:5060;user=phone SIP/2.0..Record-Route: <sip:172.17 .4.22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch =z9hG4bKbd8b.80bd0bc3.2..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK37 99992784-846..Max-Forwards: 69..Proxy-Authorization: Digest username="613", realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3ca",uri ="s ip:3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd45a",al gor ithm=md5..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..C all-ID: 3795992784-846..CSeq: 2 ACK..Contact: <sip:203@172.17.2.11:5060>..C ontent-Length: 0.... 2. ****** The call is established. The did=55f.4491d5b2. Now 3525(a)172.17.4.22 initiates call transfer. It sends re-INVITE in order to mute the call partner. The header filed Route contains did=55f.4491d5b2. U 2007/03/18 12:54:16.394546 172.17.2.14:5060 -> 172.17.4.22:5060 INVITE sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branc h=z9hG4bKb9469def476f37f2..Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992 784-846;did=55f.4491d5b2>..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae 245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Contact: <sip:3 525(a)172.17.2.14;user=phone>..Supported: replaces..Call-ID: 3795992784-846.. CSeq: 321 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70. .Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content- Type: application/sdp..Content-Length: 320....v=0..o=3525 8000 8001 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 0.0.0.0..t=0 0..m=audio 5004 RTP/AVP 0 8 18 4 99 2 9..a=sendonly..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpm ap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap:99 iLBC/8000..a=fmtp:99 mod e=20..a=rtpmap:2 G726-32/8000..a=rtpmap:9 G722/16000..a=ptime:20.. U 2007/03/18 12:54:16.396983 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.17 .2.14;branch=z9hG4bKb9469def476f37f2..From: <sip:3525@172.17.4.22:5060>;tag =61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Call -ID: 3795992784-846..CSeq: 321 INVITE..Server: ClaroSIS (Openser 1.1.1 (i38 6/linux))..Content-Length: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=3997 req_src_ip=172.17.2.14 req_src_port=5060 in_uri=sip:203@1 72.17.2.11:5060 out_uri=sip:203@172.17.2.11:5060 via_cnt==1".... 3. ****** OpenSer sends the re-INVITE to the transferee. Please note, that "did" doesn't appear in the Record-Route. As result of that, BYE for this call will no have "did" parameter and the dialog will appear as not closed. U 2007/03/18 12:54:16.397307 172.17.4.22:5060 -> 172.17.2.11:5060 INVITE sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr= on;ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK4751. 82bc6f96.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKb9469def476f37f2..Fr om: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4 .22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=phone>.. Supported: replaces..Call-ID: 3795992784-846..CSeq: 321 INVITE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE ,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Conten t-Length: 320....v=0..o=3525 8000 8001 IN IP4 172.17.2.14..s=SIP Call..c=IN IP4 0.0.0.0..t=0 0..m=audio 5004 RTP/AVP 0 8 18 4 99 2 9..a=sendonly..a=rt pmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G 723/8000..a=rtpmap:99 iLBC/8000..a=fmtp:99 mode=20..a=rtpmap:2 G726-32/8000 ..a=rtpmap:9 G722/16000..a=ptime:20.. U 2007/03/18 12:54:16.435210 172.17.2.11:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK4751.82bc6f96.0, SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKb9469def476f37f2..Record-Route: <sip: 172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Claro BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22> ;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CSeq: 321 INVITE..Conta ct: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK, INFO,REFER,NOTIFY..Content-Type: application/sdp..Content-Length: 167....v= 0..o=Claro 1174230681570 1174230681 IN IP4 172.17.2.11..s=Claro..c=IN IP4 1 72.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0..a=rtpmap:0 PCMU/8000..a=ptime:2 0..a=recvonly.. U 2007/03/18 12:54:16.436772 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKb9469def476f37f2 ..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Clar o BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CS eq: 321 INVITE..Contact: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS, CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Content-Type: application/sdp..Cont ent-Length: 167....v=0..o=Claro 1174230681570 1174230681 IN IP4 172.17.2.11 ..s=Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0..a=rtpmap:0 PCMU/8000..a=ptime:20..a=recvonly.. U 2007/03/18 12:54:16.443328 172.17.2.14:5060 -> 172.17.4.22:5060 ACK sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z 9hG4bK26b04409b72e06f9..Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55 >..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172 .17.4.22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=pho ne>..Call-ID: 3795992784-846..CSeq: 321 ACK..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIO NS,INFO,SUBSCRIBE..Content-Length: 0.... U 2007/03/18 12:54:16.445463 172.17.4.22:5060 -> 172.17.2.11:5060 ACK sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=on; ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK4751.82b c6f96.2..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK26b04409b72e06f9..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22 >;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=phone>..Cal l-ID: 3795992784-846..CSeq: 321 ACK..User-Agent: Grandstream BT110 1.0.8.16 ..Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO, SUBSCRIBE..Content-Length: 0.... U 2007/03/18 12:54:20.272078 172.17.2.14:5060 -> 172.17.4.22:5060 REFER sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branch =z9hG4bK9e02f3ad995f8245..Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b 55>..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@1 72.17.4.22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=p hone>..Supported: replaces..Refer-To: <sip:5524@172.17.4.22;user=phone>..Re ferred-By: <sip:3525@172.17.4.22;user=phone>..Call-ID: 3795992784-846..CSeq : 322 REFER..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70..Allo w: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt h: 0.... U 2007/03/18 12:54:20.275997 172.17.4.22:5060 -> 172.17.2.11:5060 REFER sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=o n;ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK1751.e 223fee7.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK9e02f3ad995f8245..Fro m: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4. 22>;tag=hssUA_3795992784-846..Contact: <sip:3525@172.17.2.14;user=phone>..S upported: replaces..Refer-To: <sip:5524@172.17.4.22;user=phone>..Referred-B y: <sip:3525@172.17.4.22;user=phone>..Call-ID: 3795992784-846..CSeq: 322 RE FER..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 69..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... U 2007/03/18 12:54:20.323290 172.17.2.11:5060 -> 172.17.4.22:5060 INVITE sip:5524@172.17.4.22;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806597784-846..Max-Forwards: 70..Allow: INVITE,OPTION S,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:2 03(a)172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22;user=ph one>..Call-ID: 3806597784-846..CSeq: 1 INVITE..Contact: <sip:203@172.17.2.1 1:5060>..Referred-By: <sip:3525@172.17.4.22;user=phone>..Content-Type: appl ication/sdp..Content-Length: 237....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s= Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpm ap:0 PCMU/8000..a=rtpmap:101 TELEPHONE-EVENT/8000..a=fmtp:101 0-16..a=rtpma p:120 BSRTCP/8000..a=fmtp:120 ver2..a=sendrecv.. U 2007/03/18 12:54:20.324935 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 172.17.2.11:506 0;branch=z9hG4bK3806597784-846..From: <sip:203@172.17.4.22>;tag=hssUA_38065 97784-846..To: <sip:5524@172.17.4.22;user=phone>;tag=329cfeaa6ded039da25ff8 cbb8668bd2.dc60..Call-ID: 3806597784-846..CSeq: 1 INVITE..Proxy-Authenticat e: Digest realm="172.17.4.22", nonce="45fd6fe80721e4ce9c9b2eef4c93a66ef88b9 f2c"..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Content-Length: 0..War ning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4001 req_src_ip=172. 17.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;user=phone out_uri=si p:5524@172.17.4.22;user=phone via_cnt==1".... U 2007/03/18 12:54:20.328647 172.17.2.11:5060 -> 172.17.4.22:5060 SIP/2.0 202 Accepted..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK1751.e223f ee7.0,SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK9e02f3ad995f8245..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..From: <sip:3525@172.17.4.22 :5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784 -846..Call-ID: 3795992784-846..CSeq: 322 REFER..Contact: <sip:203@172.17.2. 11:5060>..Content-Length: 0.... U 2007/03/18 12:54:20.330099 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 202 Accepted..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bK9e02f3ad99 5f8245..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22> ;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CSeq: 322 REFER..Contac t: <sip:203@172.17.2.11:5060>..Content-Length: 0.... U 2007/03/18 12:54:20.351529 172.17.2.11:5060 -> 172.17.4.22:5060 ACK sip:5524@172.17.4.22;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5 060;branch=z9hG4bK3806597784-846..Max-Forwards: 70..User-Agent: Claro BS320 2 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524 @172.17.4.22;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.dc60..Call -ID : 3806597784-846..CSeq: 1 ACK..Contact: <sip:203@172.17.2.11:5060>..Content -Length: 0.... U 2007/03/18 12:54:20.364969 172.17.2.11:5060 -> 172.17.4.22:5060 INVITE sip:5524@172.17.4.22;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806642784-846..Max-Forwards: 70..Allow: INVITE,OPTION S,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To : <sip:5524@172.17.4.22;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INVIT E..Contact: <sip:203@172.17.2.11:5060>..Content-Type: application/sdp..Prox y-Authorization: Digest username="613",realm="172.17.4.22",nonce="45fd6fe80 721e4ce9c9b2eef4c93a66ef88b9f2c",uri="sip:5524@172.17.4.22",response="3f b20 f7d6e67668e07b862b99412162d",algorithm=md5..Content-Length: 237....v=0..o=C laro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t=0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TELEPHONE-EVENT /8000..a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 ver2..a=sendre cv.. U 2007/03/18 12:54:20.372610 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.17 .2.11:5060;branch=z9hG4bK3806642784-846..From: <sip:203@172.17.4.22>;tag=hs sUA_3806597784-846..To: <sip:5524@172.17.4.22;user=phone>..Call-ID: 3806597 784-846..CSeq: 2 INVITE..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Con tent-Length: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4 003 req_src_ip=172.17.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;us er=phone out_uri=sip:5524@172.17.2.168:5061 via_cnt==1".... U 2007/03/18 12:54:20.372833 172.17.4.22:5060 -> 172.17.2.168:5061 INVITE sip:5524@172.17.2.168:5061 SIP/2.0..Record-Route: <sip:172.17.4.22;l r=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Via: SIP/2.0/UDP 172.17.4. 22;branch=z9hG4bK95c6.337d9f56.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch= z9hG4bK3806642784-846..Max-Forwards: 69..Allow: INVITE,OPTIONS,CANCEL,ACK,B YE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..User-Agent: Claro BS3202 A 454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524@17 2.17.4.22;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INVITE..Contact: <s ip:203@172.17.2.11:5060>..Content-Type: application/sdp..Content-Length: 23 7....v=0..o=Claro 0 1 IN IP4 172.17.2.11..s=Claro..c=IN IP4 172.17.2.11..t= 0 0..m=audio 30006 RTP/AVP 0 101 120..a=rtpmap:0 PCMU/8000..a=rtpmap:101 TE LEPHONE-EVENT/8000..a=fmtp:101 0-16..a=rtpmap:120 BSRTCP/8000..a=fmtp:120 v er2..a=sendrecv.. U 2007/03/18 12:54:20.385872 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 100 Trying..To: <sip:5524@172.17.4.22;user=phone>..From: <sip:203@1 72.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806597784-846..CSeq: 2 INVI TE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95c6.337d9f56.0..Via: SIP/2. 0/UDP 172.17.2.11:5060;branch=z9hG4bK3806642784-846..Record-Route: <sip:172 .17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Server: Linksys/ PAP2-3.1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.393629 172.17.2.11:5060 -> 172.17.4.22:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806672784-846..Route: <sip:172.17.4.22;lr=on;ftag=hss UA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Authorization: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342c c574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b864a 4ab 937d17f42c7cd45a",algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK ,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..C all-ID: 3795992784-846..CSeq: 3 NOTIFY..Contact: <sip:203@172.17.2.11:5060> ..Event: refer..Subscription-State: active;expires=60..Content-Type: messag e/sipfrag..Content-Length: 530....SIP/2.0 100 trying -- your call is import ant to us..Via: SIP/2.0/UDP 172.17.2.11:5060;received=172.17.4.22;branch=z9 hG4bK3806642784-846..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846.. To: <sip:5524@172.17.4.22;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INV ITE..Server: ClaroSIS (Openser 1.1.1 (i386/linux))..Content-Length: 0..Warn ing: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4003 req_src_ip=172.1 7.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;user=phone out_uri=sip :5524@172.17.2.168:5061 via_cnt==1".... U 2007/03/18 12:54:20.396980 172.17.4.22:5060 -> 172.17.2.14:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip:172.17.4 .22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z 9hG4bKcd8b.81aaa3f3.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806 672784-846..Max-Forwards: 69..Proxy-Authorization: Digest username="613",re alm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3ca",uri=" sip :3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd45a",algo rit hm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Suppo rted: replaces..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: < sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..C Seq: 3 NOTIFY..Contact: <sip:203@172.17.2.11:5060>..Event: refer..Subscript ion-State: active;expires=60..Content-Type: message/sipfrag..Content-Length : 530....SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/U DP 172.17.2.11:5060;received=172.17.4.22;branch=z9hG4bK3806642784-846..From : <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22 ;user=phone>..Call-ID: 3806597784-846..CSeq: 2 INVITE..Server: ClaroSIS (Op enser 1.1.1 (i386/linux))..Content-Length: 0..Warning: 392 172.17.4.22:5060 "Noisy feedback tells: pid=4003 req_src_ip=172.17.2.11 req_src_port=5060 in_uri=sip:5524@172.17.4.22;user=phone out_uri=sip:5524@172.17.2.168:5061 v ia_cnt==1".... U 2007/03/18 12:54:20.401630 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 180 Ringing..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff 0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806 597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95c6 .337d9f56.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806642784-846 ..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c 5bd92>..Server: Linksys/PAP2-3.1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.403183 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 180 Ringing..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff 0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806 597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4b K3806642784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784 -846;did=0e3.7c5bd92>..Server: Linksys/PAP2-3.1.3(LS)..Content-Length: 0... . U 2007/03/18 12:54:20.403878 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKcd8b.81aaa3f3.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806672784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.1 7.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645 575ae245b55..Call-ID: 3795992784-846..CSeq: 3 NOTIFY..User-Agent: Grandstre am BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=phone>..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Supported: replaces.. Content-Length: 0.... U 2007/03/18 12:54:20.407617 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806672784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22: 5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 3 NOTIFY..User-A gent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14:5060;user= phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..S upported: replaces..Content-Length: 0.... U 2007/03/18 12:54:20.429109 172.17.2.11:5060 -> 172.17.4.22:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Via: SIP/2.0/UDP 172.17.2.1 1:5060;branch=z9hG4bK3806702784-846..Route: <sip:172.17.4.22;lr=on;ftag=hss UA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Authorization: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342c c574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156ea61b864a 4ab 937d17f42c7cd45a",algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK ,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:203@172.17.4.22>;tag=hs sUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..C all-ID: 3795992784-846..CSeq: 4 NOTIFY..Contact: <sip:203@172.17.2.11:5060> ..Event: refer..Subscription-State: active;expires=60..Content-Type: messag e/sipfrag..Content-Length: 397....SIP/2.0 180 Ringing..To: <sip:5524@172.17 .4.22;user=phone>;tag=8cb5cce9fbff0a22i1..From: <sip:203@172.17.4.22>;tag=h ssUA_3806597784-846..Call-ID: 3806597784-846..CSeq: 2 INVITE..Via: SIP/2.0/ UDP 172.17.2.11:5060;received=172.17.4.22;branch=z9hG4bK3806642784-846..Rec ord-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92 >..Server: Linksys/PAP2-3.1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.431776 172.17.4.22:5060 -> 172.17.2.14:5060 NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip:172.17.4 .22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z 9hG4bK9d8b.47c7ce13.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806 702784-846..Max-Forwards: 69..Proxy-Authorization: Digest username="613",re alm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3ca",uri=" sip :3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd45a",algo rit hm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Suppo rted: replaces..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: < sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..C Seq: 4 NOTIFY..Contact: <sip:203@172.17.2.11:5060>..Event: refer..Subscript ion-State: active;expires=60..Content-Type: message/sipfrag..Content-Length : 397....SIP/2.0 180 Ringing..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb 5cce9fbff0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call -ID: 3806597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.2.11:5060;rece ived=172.17.4.22;branch=z9hG4bK3806642784-846..Record-Route: <sip:172.17.4. 22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Server: Linksys/PAP2-3 .1.3(LS)..Content-Length: 0.... U 2007/03/18 12:54:20.438291 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK9d8b.47c7ce13.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806702784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.1 7.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645 575ae245b55..Call-ID: 3795992784-846..CSeq: 4 NOTIFY..User-Agent: Grandstre am BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=phone>..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Supported: replaces.. Content-Length: 0.... U 2007/03/18 12:54:20.439954 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806702784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22: 5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 4 NOTIFY..User-A gent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14:5060;user= phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..S upported: replaces..Content-Length: 0.... U 2007/03/18 12:54:23.312128 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95c6.337d 9f56.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806642784-846..Rec ord-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92 >..Contact: 5524 <sip:5524@172.17.2.168:5061>..Server: Linksys/PAP2-3.1.3(L S)..Content-Length: 210..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPT IONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17.2.168..t=0 0.. m=audio 16388 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-e vent/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 2007/03/18 12:54:23.313878 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3806 642784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846; did=0e3.7c5bd92>..Contact: 5524 <sip:5524@172.17.2.168:5061>..Server: Links ys/PAP2-3.1.3(LS)..Content-Length: 210..Allow: ACK, BYE, CANCEL, INFO, INVI TE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/ sdp....v=0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17 .2.168..t=0 0..m=audio 16388 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap: 101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 2007/03/18 12:54:23.355581 172.17.2.11:5060 -> 172.17.4.22:5060 27646@0:1480 ........NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Via: SIP/2.0/UDP 17 2.17.2.11:5060;branch=z9hG4bK3809632784-846..Route: <sip:172.17.4.22;lr=on; ftag=hssUA_3795992784-846;did=55f.4491d5b2>..Max-Forwards: 70..Proxy-Author ization: Digest username="613",realm="172.17.4.22",nonce="45fd6fdd9e385352f 9c63342cc574c454939f3ca",uri="sip:3525@172.17.4.22:5060",response="7156e a61 b864a4ab937d17f42c7cd45a",algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,B YE,PRACK,INFO,REFER,NOTIFY..Supported: replaces..From: <sip:203@172.17.4.22 >;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae2 45b55..Call-ID: 3795992784-846..CSeq: 5 NOTIFY..Contact: <sip:203@172.17.2. 11:5060>..Event: refer..Subscription-State: terminated;reason=noresource..C ontent-Type: message/sipfrag..Content-Length: 756....SIP/2.0 200 OK..To: <s ip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i1..From: <sip:203@172. 17.4.22>;tag=hssUA_3806597784-846..Call-ID: 3806597784-846..CSeq: 2 INVITE. .Via: SIP/2.0/UDP 172.17.2.11:5060;received=172.17.4.22;branch=z9hG4bK38066 42784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;d id=0e3.7c5bd92>..Contact: 5524 <sip:5524@172.17.2.168:5061>..Server: Linksy s/PAP2-3.1.3(LS)..Content-Length: 210..Allow: ACK,BYE,CANCEL,INFO,INVITE,NO TIFY,OPTIONS,REFER..Supported: x-sipura..Content-Type: application/sdp....v =0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17.2.168.. t=0 0..m=audio 16388 RTP/AVP 0 101..a=rtpmap:0 PCMU/800 U 2007/03/18 12:54:23.361462 172.17.4.22:5060 -> 172.17.2.14:5060 50085@0:1480 .....I..NOTIFY sip:3525@172.17.2.14;user=phone SIP/2.0..Record-Route: <sip: 172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..Via: SIP/2.0/UDP 172.17.4.22; branch=z9hG4bKad8b.016fb321.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9h G4bK3809632784-846..Max-Forwards: 69..Proxy-Authorization: Digest username= "613",realm="172.17.4.22",nonce="45fd6fdd9e385352f9c63342cc574c454939f3c a", uri="sip:3525@172.17.4.22:5060",response="7156ea61b864a4ab937d17f42c7cd4 5a" ,algorithm=md5..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INFO,REFER,NOTIF Y..Supported: replaces..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-84 6..To: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..Call-ID: 379599278 4-846..CSeq: 5 NOTIFY..Contact: <sip:203@172.17.2.11:5060>..Event: refer..S ubscription-State: terminated;reason=noresource..Content-Type: message/sipf rag..Content-Length: 756....SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user= phone>;tag=8cb5cce9fbff0a22i1..From: <sip:203@172.17.4.22>;tag=hssUA_380659 7784-846..Call-ID: 3806597784-846..CSeq: 2 INVITE..Via: SIP/2.0/UDP 172.17. 2.11:5060;received=172.17.4.22;branch=z9hG4bK3806642784-846..Record-Rout e: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846;did=0e3.7c5bd92>..Conta ct: 5524 <sip:5524@172.17.2.168:5061>..Server: Linksys/PAP2-3.1.3(LS)..Content -Length: 210..Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER..Suppo rted: x-sipura..Content-Type: application/sdp....v=0..o=- 23586680 23586680 IN IP4 172.17.2.168..s=-..c=IN IP4 172.17.2.168..t=0 0 U 2007/03/18 12:54:23.371944 172.17.2.11:5060 -> 172.17.4.22:5060 ACK sip:5524@172.17.2.168:5061 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5060;b ranch=z9hG4bK3809647784-846..Route: <sip:172.17.4.22;lr=on;ftag=hssUA_38065 97784-846;did=0e3.7c5bd92>..Max-Forwards: 70..Proxy-Authorization: Digest u sername="613",realm="172.17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66 ef8 8b9f2c",uri="sip:5524@172.17.4.22",response="3fb20f7d6e67668e07b862b9941 216 2d",algorithm=md5..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.2 2>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5 cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 2 ACK..Contact: <sip:203@172 .17.2.11:5060>..Content-Length: 0.... U 2007/03/18 12:54:23.372334 172.17.2.14:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKad8b.016fb321.0. .Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3809632784-846..Record-Rou te: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.1 7.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22:5060>;tag=61645 575ae245b55..Call-ID: 3795992784-846..CSeq: 5 NOTIFY..User-Agent: Grandstre am BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14;user=phone>..Allow: INVIT E,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Supported: replaces.. Content-Length: 0.... U 2007/03/18 12:54:23.373917 172.17.4.22:5060 -> 172.17.2.168:5061 ACK sip:5524@172.17.2.168:5061 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=o n;ftag=hssUA_3806597784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK95 c6.337d9f56.2..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3809647784-8 46..Max-Forwards: 69..Proxy-Authorization: Digest username="613",realm="172 .17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66ef88b9f2c",uri="sip:5524 @17 2.17.4.22",response="3fb20f7d6e67668e07b862b99412162d",algorithm=md5..Us er- Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784- 846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 2 ACK..Contact: <sip:203@172.17.2.11:5060>..Content- Length: 0.... U 2007/03/18 12:54:23.375179 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3809632784- 846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3795992784-846>..From: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..To: <sip:3525@172.17.4.22: 5060>;tag=61645575ae245b55..Call-ID: 3795992784-846..CSeq: 5 NOTIFY..User-A gent: Grandstream BT110 1.0.8.16..Contact: <sip:3525@172.17.2.14:5060;user= phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..S upported: replaces..Content-Length: 0.... 4. ****** The message BYE doesn't have the did parameter. U 2007/03/18 12:54:23.490685 172.17.2.14:5060 -> 172.17.4.22:5060 BYE sip:203@172.17.2.11:5060 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z 9hG4bKa419f2aaf11a7b40..Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55 >..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172 .17.4.22>;tag=hssUA_3795992784-846..Supported: replaces..Call-ID: 379599278 4-846..CSeq: 323 BYE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... U 2007/03/18 12:54:23.494656 172.17.4.22:5060 -> 172.17.2.11:5060 BYE sip:203@172.17.2.11:5060 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=on; ftag=61645575ae245b55>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK2751.609 47d11.0..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKa419f2aaf11a7b40..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22 >;tag=hssUA_3795992784-846..Supported: replaces..Call-ID: 3795992784-846..C Seq: 323 BYE..User-Agent: Grandstream BT110 1.0.8.16..Max-Forwards: 69..All ow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... U 2007/03/18 12:54:23.517688 172.17.2.11:5060 -> 172.17.4.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bK2751.60947d11.0, SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKa419f2aaf11a7b40..Record-Route: <sip: 172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Claro BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22> ;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CSeq: 323 BYE..Contact: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS,CANCEL,ACK,BYE,PRACK,INF O,REFER,NOTIFY..Content-Length: 0.... U 2007/03/18 12:54:23.519138 172.17.4.22:5060 -> 172.17.2.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.17.2.14;branch=z9hG4bKa419f2aaf11a7b40 ..Record-Route: <sip:172.17.4.22;lr=on;ftag=61645575ae245b55>..Server: Clar o BS3202 A454..From: <sip:3525@172.17.4.22:5060>;tag=61645575ae245b55..To: <sip:203@172.17.4.22>;tag=hssUA_3795992784-846..Call-ID: 3795992784-846..CS eq: 323 BYE..Contact: <sip:203@172.17.2.11:5060>..Allow: INVITE,OPTIONS,CAN CEL,ACK,BYE,PRACK,INFO,REFER,NOTIFY..Content-Length: 0.... U 2007/03/18 12:54:29.780666 172.17.2.11:5060 -> 172.17.4.22:5060 BYE sip:5524@172.17.2.168:5061 SIP/2.0..Via: SIP/2.0/UDP 172.17.2.11:5060;b ranch=z9hG4bK3816062784-846..Route: <sip:172.17.4.22;lr=on;ftag=hssUA_38065 97784-846;did=0e3.7c5bd92>..Max-Forwards: 70..Proxy-Authorization: Digest u sername="613",realm="172.17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66 ef8 8b9f2c",uri="sip:5524@172.17.4.22",response="3fb20f7d6e67668e07b862b9941 216 2d",algorithm=md5..User-Agent: Claro BS3202 A454..From: <sip:203@172.17.4.2 2>;tag=hssUA_3806597784-846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5 cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 3 BYE..Content-Length: 0.... U 2007/03/18 12:54:29.784698 172.17.4.22:5060 -> 172.17.2.168:5061 BYE sip:5524@172.17.2.168:5061 SIP/2.0..Record-Route: <sip:172.17.4.22;lr=o n;ftag=hssUA_3806597784-846>..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKa5 c6.704043c7.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3816062784-8 46..Max-Forwards: 69..Proxy-Authorization: Digest username="613",realm="172 .17.4.22",nonce="45fd6fe80721e4ce9c9b2eef4c93a66ef88b9f2c",uri="sip:5524 @17 2.17.4.22",response="3fb20f7d6e67668e07b862b99412162d",algorithm=md5..Us er- Agent: Claro BS3202 A454..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784- 846..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i1..Call-ID: 3806597784-846..CSeq: 3 BYE..Content-Length: 0.... U 2007/03/18 12:54:29.804689 172.17.2.168:5061 -> 172.17.4.22:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 3 BYE..Via: SIP/2.0/UDP 172.17.4.22;branch=z9hG4bKa5c6.704043c 7.0..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3816062784-846..Record -Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846>..Server: Linksys/ PAP2-3.1.3(LS)..P-RTP-Stat: PS=146,OS=34640,PR=303,OR=48162,PL=1,JI=0,LA=10 ,DU=6,EN=G711u,DE=G711u..Content-Length: 0.... U 2007/03/18 12:54:29.806174 172.17.4.22:5060 -> 172.17.2.11:5060 SIP/2.0 200 OK..To: <sip:5524@172.17.4.22;user=phone>;tag=8cb5cce9fbff0a22i 1..From: <sip:203@172.17.4.22>;tag=hssUA_3806597784-846..Call-ID: 380659778 4-846..CSeq: 3 BYE..Via: SIP/2.0/UDP 172.17.2.11:5060;branch=z9hG4bK3816062 784-846..Record-Route: <sip:172.17.4.22;lr=on;ftag=hssUA_3806597784-846>..S erver: Linksys/PAP2-3.1.3(LS)..P-RTP-Stat: PS=146,OS=34640,PR=303,OR=48162, PL=1,JI=0,LA=10,DU=6,EN=G711u,DE=G711u..Content-Length: 0.... As you can see, the dialog (callid=3795992784-846) still exists... /usr/local/sbin/openserctl fifo dlg_list 200 OK hash=3925, label=727521604, ptr=0x405c2d78 state=3, timeout=236218 callid='3795992784-846' from uri='sip:203@172.17.4.22', tag='hssUA_3795992784-846' to uri='sip:3525@172.17.4.22:5060', tag='61645575ae245b55' Thnk you in advance, Leonid Fainshtein -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, March 16, 2007 2:51 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Problem with dialog module Hi Leonid, what do you mean by "removes"?. When sequential requests are processed, the Route header corresponding to your server will be completely removed. regards, bogdan Leonid Fainshtein wrote: > Hi, > I see that when Openser 1.1 receives re-INVITE it removes parameter > "did=" from header field Route before sending it to the second call > party. > Any idea? > > Thank you in advance, > Leonid > > > ******************************************** > This outgoing mail message was scanned by McAfee GroupShield Engine > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 2

[Serusers] SER with TLS
by Katty Xiong 04 Apr '07

04 Apr '07

I am using SER ottendorf with TLS protocol and have the following issues. Does anybody experience similar problems? SER cannot run with the following setup in the configuration file: (I follow this link to setup key and certificate: http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README…) modparam("tls", "private_key", "cakey.pem") modparam("tls", "certificate", "cacert.pem") modparam("tls", "ca_list", "calist.pem") modparam("tls", "cipher_list", "HIGH"); With the last line commented out: #modparam("tls", "cipher_list", "HIGH"); SER can start, but the tls connection cannot be established. Network trace shows SER does not responde to ClientHello sent by client. Thanks, Joy ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

3 9

[Users] response message content
by Victor Gamov 03 Apr '07

03 Apr '07

Hi all! Can somebody explain me how I can fill response with my useful headers? For example, I need to send 305 "Use proxy" response with Contact headers. Is it possible? If so, how can I populate it? Thanks! -- CU, Victor Gamov

3 5

Re: [Serusers] How to disable RTP proxy connection in
by Amanda Mattiuz 03 Apr '07

03 Apr '07

What happens if I disable rtpproxy but still load the nathelper? Can I still solve NAT problems with that?? -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

3 2

[Users] RFC 3428
by Helmut Kuper 03 Apr '07

03 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have a question about openser: Does openser support RFC 3428 (Instant Messaging (IM))? If so, How do I have to configure openser to relay SIP method MESSAGE ? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGERzN4tZeNddg3dwRAkNgAJ9t+ffNXs8+tiISFPEZlSyf7A5m2ACfSzJM rKsinfXGvAWBBW6mMSO9a8Y= =ixv4 -----END PGP SIGNATURE-----

3 3

Re: Fwd: [Users] SNMPstats module not returning data
by Terry Robert 03 Apr '07

03 Apr '07

I experienced the same problem with the SNMPstats module. If you run openser in foreground mode (fork=no), the module doesn't initialize. Running in background mode fixes the problem.

3 3

Re: [Users] Openser fails when compiled on Solaris 64bit
by Sergio Gutierrez 03 Apr '07

03 Apr '07

Hi Daniel. This is the backtrace of the core. Thanks #0 0xffffffff7d90ee9c in preload_udomain (_c=0x10026b7d800, _d=0x0) at /home/operador/openser-1.2.0-notls/modules/usrloc//udomain.c:404 #1 0xffffffff7d9131c4 in child_init (_rank=1344858528) at /home/operador/openser-1.2.0-notls/modules/usrloc//ul_mod.c:344 #2 0x0000000100082500 in init_mod_child (m=0x1, rank=1, type=0x1000f41b0 "CHILD") at /home/operador/openser-1.2.0-notls//sr_module.c:400 #3 0x0000000100082460 in init_mod_child (m=0x1, rank=1, type=0x1000f41b0 "CHILD") at /home/operador/openser-1.2.0-notls//sr_module.c:394 #4 0x0000000100082460 in init_mod_child (m=0x1, rank=1, type=0x1000f41b0 "CHILD") at /home/operador/openser-1.2.0-notls//sr_module.c:394 #5 0x0000000100082770 in init_child (rank=1) at /home/operador/openser- 1.2.0-notls//sr_module.c:394 #6 0x0000000100032414 in main_loop () at /home/operador/openser-1.2.0-notls //main.c:724 #7 0x00000001000345a4 in main (argc=2, argv=0xffffff7e7007a000) at /home/operador/openser-1.2.0-notls//main.c:1399 On 3/26/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > Hello, > > can you send a gdb backtrace :-D -- I cannot read adb backtrace. > > Cheers, > Daniel > > > On 03/26/07 19:42, Sergio Gutierrez wrote: > > Hi Daniel. > > > > This is the backtrace, obtained with adb: > > > > usrloc.so`preload_udomain+0x4cc(1, ffffffff7228e9f0, 100215a00, > > ffffffff7d919e58, ffffffffffefe3f8, 0) > > 0xffffffff7d9131bc(2, ffffffff7da1c9b0, ffffffff7d91a170, > > ffffffff7228e9a0, ffffffff7da1b0a8, ffffffffffeff0c8) > > init_mod_child+0xd8(100269930, 1, 1000f41b0, 1000f40f0, 1, 100269870) > > init_mod_child+0x38(100269bb0, 1, 1000f41b0, 1000f40f0, 1, 1002699f0) > > init_mod_child+0x38(100269d30, 1, 1000f41b0, 1000f40f0, 1, 100269c70) > > init_child+0xa8(1, 100269df0, 1, 1000f4168, 10020d000, 100269eb0) > > main_loop+0xf34(0, 8c, 1000eefd8, ffffffff720083c4, 1002694b0, 0) > > main+0x1e10(100215, 9, 0, 10020d000, 1000ef5b8, 10020d000) > > _start+0x7c(0, 0, 0, 0, 0, 0) > > > > On 3/26/07, *Sergio Gutierrez* <saguti(a)gmail.com > > <mailto:saguti@gmail.com>> wrote: > > > > Hi Daniel. > > > > I used 512 for -m parameter. > > > > Would you like the backtrace of the core file? > > > > Thanks. > > > > > > On 3/26/07, *Daniel-Constantin Mierla* < daniel(a)voice-system.ro > > <mailto:daniel@voice-system.ro>> wrote: > > > > Hello, > > > > could you send a backtrace? What was the value for parameter -m? > > > > Cheers, > > Daniel > > > > > > On 03/26/07 19:29, Sergio Gutierrez wrote: > > > Hi > > > > > > I am trying to run Openser compiled on 64 bits on a SPARC > > Machine > > > running Solaris 10. > > > > > > When I create a simple configuration file for testing radius > > > integration, Openser starts to consume the whole memory > > reservation > > > (-m parameter) and fails with segmentation fault error. > > > > > > after several tests, I have found that the error is caused by > > save() > > > function (registrar module). > > > > > > This is the main route my configuration file: > > > > > > route { > > > if(method=="REGISTER") > > > { > > > if(!radius_www_authorize("")) > > > { > > > www_challenge("", "0"); > > > return; > > > }; > > > > > > if(!save("location")) > > > { > > > sl_reply_error(); > > > }; > > > return; > > > > > > } > > > else > > > { > > > } > > > } > > > > > > Thanks in advance for your help. > > > > > > Sergio G. > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org <mailto:Users@openser.org> > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > >

3 10

[Users] Module Permission (register.allow and register.deny)
by Andrew And 03 Apr '07

03 Apr '07

In http://www.openser.org/docs/modules/1.2.x/permissions.html don´t have example with files register.allow and register.deny. Does someone has one example with rules these files?

1 0

[Users] does openser support QoS ?
by yanlin 03 Apr '07

03 Apr '07

Hi, does openser support some kind of QoS ? like, band width management, when there has no more usable bandwidth, then new INVITE should be rejected. so far, i know that dialog module has a bit dialog awareness. Any clues? thanks. yan lin

4 3

[Users] Serial forking broken wrt dst_uri/path?
by Frey, Daniel (external) 03 Apr '07

03 Apr '07

Hi, I tried to use append_branch() with dst_uri and path parameters set (and using set_dst_uri() / set_path_vector() for the first message). It works well as long as only parallel forking is involved, but when I use serialize_branches and next_branches and serial forking is required, the information given in dst_uri and path is discarded. This also seems obvious when reading the code from serialize.c, which simply doesn't store it on the contacts[n]-structs. I could add it to the sturct, but I'm at a loss how to store/retrieve it from the AVP. Are there any plans to fix this? Or any help on how I can fix it myself? Regards, Daniel

2 1

[Users] expires in Register message
by Dimo 03 Apr '07

03 Apr '07

Hi guys, Congratulations on the recent release, I will be testing it soon. I wonder if you can help me with a problem. I am trying to set UAs with a certain user-agent header to have different max_expires than all the rest. I am trying to override the default min_expires and max_expires values from within the config, after the definitions. So far i cannot do it - the modparam("registrar", "min_expires", xx) does not seem to accept an AVP for the xx part. Do you know if it is possible to change these timers per client from the code? Do they accept AVPs or is there any other way i can make it work? Thanks in advance Dimo

2 1

[Users] RTP without proxying, Linksys 3102
by Amanda Mattiuz 03 Apr '07

03 Apr '07

Hello Bill... it would be great if you could send me the cfg file you used.... -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

1 0

[Serusers] mail client for SER UA
by Meidiana Ten 03 Apr '07

03 Apr '07

Hi there, I'm setting up voicemail server using SER. Do I need to setup mail client in client machine so it can access mailbox in server? If so, please tell me the software and how to configure the mail client. Thanx Regards, Meidiana ____________________________________________________________________________________ Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/

1 0

RE: [Serusers] global contact list
by Barta János 03 Apr '07

03 Apr '07

Hello! Thanks for your answer. I'm using the standard config file (http://ftp.iptel.org/pub/ser/presence/cfg/full-no-failover.cfg), with a little modification, because I'm using radius authentication, accounting and media proxy. loadmodule "/opt/lib/ser/modules/acc_radius.so" loadmodule "/opt/lib/ser/modules/mediaproxy.so" loadmodule "/opt/lib/ser/modules/auth_radius.so" modparam("acc_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") modparam("acc_radius", "log_flag", 1) modparam("acc_radius", "log_missed_flag", 2) modparam("rls", "db_url", "mysql://yyyyyyy:xxxxxx@10.22.1.222/ser_pa") modparam("domain|uri_db|auth_db|usrloc|msilo", "db_url", "mysql://yyyyyyy:xxxxxx@10.22.1.222/ser_pa") if ((method=="INVITE") or (method=="BYE") or (method=="CANCEL")) { setflag(1); } if (!radius_www_authorize("test.intra")) { www_challenge("test.intra", "0"); break; }; handle_rls_subscription("1"); 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(); } ----------------- I made a capture with Ethereal, and I can see that eyeBeam sent the "Supported: eventlist" field. I inserted the lists of both type: Global-list: Message Header Via: SIP/2.0/UDP 10.22.2.134:9510;branch=z9hG4bK-d87543-dd118615e6316f41-1--d87543- Max-Forwards: 70 Contact: <sip:12345@10.22.2.134:9510> To: <sip:bartaj-list@test.intra> SIP to address: sip:bartaj-list@test.intra From: "12345"<sip:12345@test.intra>;tag=0a48df34 SIP Display info: "12345" SIP from address: sip:12345@test.intra SIP tag: 0a48df34 Call-ID: ee79a62abd227204@S2FhbGlhLXhwLndlc3RlbC53ZXN0ZWw5MDAuaHU. CSeq: 1 SUBSCRIBE Expires: 60 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: CounterPath eyeBeam Event: presence Content-Length: 0 User-list: Message Header Via: SIP/2.0/UDP 10.22.2.134:9510;branch=z9hG4bK-d87543-6300f275d049656e-1--d87543- Max-Forwards: 70 Contact: <sip:12345@10.22.2.134:9510> To: <sip:12345@test.intra> SIP to address: sip:12345@test.intra From: "12345"<sip:12345@test.intra>;tag=cb3e2c5c SIP Display info: "12345" SIP from address: sip:12345@test.intra SIP tag: cb3e2c5c Call-ID: a46e4b35e4754144@S2FhbGlhLXhwLndlc3RlbC53ZXN0ZWw5MDAuaHU. CSeq: 1 SUBSCRIBE Expires: 3600 Accept: message/external-body Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: CounterPath eyeBeam Event: sip-profile;profile-type=application;app-id=resource-lists Content-Length: 0 The second case is working, but the first one is not. Is this maybe an EyeBeam problem? Which version of EyeBeam works fine (or which other client can I test it with)? And I have another question. What can I do with those clients, that does not support the presence information (can I do something anyway?). For example: I have a "Cisco Call Manager Express" box, and some Cisco IP Phones, that reach other PC clients through the SER. The Cisco makes the registration (and reregistration) with the phone's identifiers too into the SER (but the presence information is not handled by the Cisco), and therefore if a PC client would like to add the Cisco phone into his buddy list, the presence won't be working. ...Record(0x405cc588)... domain: 'location' aor : '101$' ~~~Contact(0x405cc5c8)~~~ domain : 'location' aor : '101$' Contact : 'sip:101$@10.22.2.233:5060' Expires : Expired q : Call-ID : '9C5717E5-D7D211DB-804EA958-71810899' CSeq : 4103 User-Agent: 'Cisco-SIPGateway/IOS-12.x' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0x405cc698)... domain: 'location' aor : '100$' ~~~Contact(0x405cc6d8)~~~ domain : 'location' aor : '100$' Contact : 'sip:100$@10.22.2.233:5060' Expires : Expired q : Call-ID : '9C5717E5-D7D211DB-804DA958-71810899' CSeq : 1364 User-Agent: 'Cisco-SIPGateway/IOS-12.x' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... Is it possible, that the SER (in these cases) simulates a presence client (if the user registration is active, then the user is online, anyway offline)? Thanks, Jani -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Wednesday, March 28, 2007 3:46 PM To: Barta János Cc: serusers(a)iptel.org Subject: Re: [Serusers] global contact list Hi, please can you send here your config file? By the way, in some EyeBeam versions was not possible to use "Subscribe to contact list" feature because in such cases EyeBeam didn't send header "Supported: eventlist". Vaclav On Tue, Mar 27, 2007 at 08:54:15AM +0200, Barta János wrote: > Hello! > > I have a little problem with the (global) contact list subscription > of the presence, please help me. > I installed the presence server (0.10.99-dev35-pa-4.1), and it works > without any problems. > But I can not use the "Subscribe to contact list" of the eyeBeam. (The > user subscribe list works fine.) I typed the next content into this > field (of the eyeBeam): > > sip:bartaj-list@test.intra > > > and my config file contains: > > sip:/var/xcap-root/rls-services/global# 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> > > </rls-services> > > But I can see only the user subscribe list, and can't see the global > one. > Why? > Can anybody help me? > Thx, > Jani > _____________________________________________________________________________________________________ Jean Michel Jarre: Téo és Téa, az új, ünnepi album 1990 Forintért, kizárólag a T-Online Zeneáruházban! http://zenearuhaz.t-online.hu/index.php?m=info&albumid=30593&sp=1&sty=21

2 1

[Users] [resolv] radius problem in 1.2
by Bodin Bruno 03 Apr '07

03 Apr '07

Hello, During few days I've a problem with openser 1.2 and radius authentication but I just solve it ! This is a little note about my method. *Please, you're welcome to check and comment it for a better solution.* *Issue* : Radius authentication fail for openser 1.2 but not for 1.1.1 version with the same config. *Description* : freeradius server sucess (Sending Access-Accept in log) but it's not good in openser log. *Search method* : analyse of differences between 1.1.1 and 1.2 version >lsof | grep radius | grep -v radiusd /_openser 1.1.1_/ : openser 6226 root mem REG 8,2 50984 1333396 /usr/lib64/libradiusclient-ng.so.2.0.0 openser 6226 root mem REG 8,2 71642 30076344 /opt/openser-1.1.1/lib64/openser/modules/auth_radius.so openser 6226 root mem REG 8,2 41637 30076357 /opt/openser-1.1.1/lib64/openser/modules/group_radius.so openser 6226 root mem REG 8,2 43683 30076346 /opt/openser-1.1.1/lib64/openser/modules/avp_radius.so openser 6226 root mem REG 8,2 38171 30389824 /opt/openser-1.1.1/lib64/openser/modules/uri_radius.so _/openser1.2 :/_ openser 6299 root mem REG 8,2 50952 1333316 /usr/lib64/libfreeradius-client.so.2.0.0 openser 6299 root mem REG 8,2 76493 30356347 /opt/openser-1.2/lib64/openser/modules/auth_radius.so openser 6299 root mem REG 8,2 41769 30356361 /opt/openser-1.2/lib64/openser/modules/group_radius.so openser 6299 root mem REG 8,2 44923 30356349 /opt/openser-1.2/lib64/openser/modules/avp_radius.so openser 6299 root mem REG 8,2 39411 30076331 /opt/openser-1.2/lib64/openser/modules/uri_radius.so *openser 1.2.0 use libfreeradius-client.so.2.0.0 openser 1.1.1 use libradiusclient-ng.so.2.0.0 * * my solve method :* >cd /usr/lib64/ >mv libfreeradius-client.so.2.0.0 libfreeradius-client.so.2.0.0.old >ln -s libradiusclient-ng.so.2.0.0 libfreeradius-client.so.2.0.0

2 1

[Users] DLG: BYE not detected because of route reccord
by Michel Bensoussan 03 Apr '07

03 Apr '07

Hello The dialog module checks SIP messages only if the OpenSER in on record-route header in function dlg_onroute(). When the callee in registered on another SIP Server and the callee sends BYE, the OpenSER is not on record-route header. In this case, the dialog module doesn't detect end of call. Is there a way to callback the dlg_onroute() function not only in case of record-route? The configuration is as following: - Tel 101 registered to 192.168.13.8 (siproxd) (URI: 101(a)192.168.13.8) (IP: 192.168.13.101) - Tel 155 registered to 192.168.13.86 (OpenSER) (URI: 155(a)192.168.13.86) (IP: 192.168.13.155) - 155 is the caller and 101 is the callee. Attached, message capture (ethereal) when the callee (101) terminate the call. Thanks. Regards, Michel.

2 5

[Users] BUG in TM
by Sam Lee 03 Apr '07

03 Apr '07

Hey guys, I'm getting BUG:tm:run_failure_handlers: stack level is 1!! in my syslog. It is generating when i place a call out to PSTN. Any ideas ? Sam

2 5

Re: [Users] Module UAC to authentication and redirect calls
by Bogdan-Andrei Iancu 03 Apr '07

03 Apr '07

See the "credential" module parameter: http://www.openser.org/docs/modules/1.2.x/uac.html#AEN106 regards, bogdan Thiago Maluf wrote: > what`s parameters I will use to do this authentication? > > 2007/3/2, Bogdan-Andrei Iancu < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi, > > try something like: > > failure_route[1] { > xlog("-----------$T_reply_code ---------\n"); > if (t_check_status("401|407")) { > log(1,"doing auth\n"); > if (uac_auth()) { > append_branch(); > t_relay(); > } > } > } > > regards, > bogdan > > Thiago Maluf wrote: > > Hi List, > > I need config one module UAC to do authentication and redirect my > > calls to other Voip Provider. > > Somebody already did it? If can tell me what functions you used. > > Thanks in advanced! > > Thiago Maluf. > > > > ---------------------------------------------------------------- > > THIAGO MALUF RESENDE > > Consultor Voip e Programador WEB (Voip Developer and Web Developer) > > e-mail: malufrj(a)gmail.com <mailto:malufrj@gmail.com> <mailto: > malufrj(a)gmail.com <mailto:malufrj@gmail.com>> > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > ---------------------------------------------------------------- > THIAGO MALUF RESENDE > Consultor Voip e Programador WEB (Voip Developer and Web Developer) > Tel: +55 21 86042100 > e-mail: malufrj(a)gmail.com <mailto:malufrj@gmail.com>

1 0

Re: [Users] 3xx requests and t_relay
by Bogdan-Andrei Iancu 03 Apr '07

03 Apr '07

Hi Shane, please post the network trace from the the server and the full debug log of openser (use debug=9). regards, bogdan Shane Burrell wrote: > I used that as an example. On the sip side I see the 302 come back and the > openser ack it. Then I never see the new call go out. The user gets a busy > or not avaible message from the switch. In openser I have some logging and > it seems to start a new route after the ACK but bombs somewhere. > > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 28, 2007 12:54 PM > To: Shane Burrell; users openser.org > Subject: Re: [Users] 3xx requests and t_relay > > Hi Shane, > > have you followed this ex: > http://www.openser.org/docs/modules/1.2.x/uac_redirect.html#AEN251 > > can you be more specific about the errors you see? > > regards, > bogdan > > Shane Burrell wrote: > >> I am calling the t_relay after getting the contacts. However internally I >> see in openser it makes it back to the route and then dies in loose route. >> I assumed there was something I need to check to allow the new call to be >> processed. >> >> -----Original Message----- >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >> Sent: Wednesday, February 28, 2007 12:26 PM >> To: Shane Burrell >> Cc: users(a)openser.org >> Subject: Re: [Users] 3xx requests and t_relay >> >> Hi, >> >> let me see if I get it right...you get a 3xx reply and try to generate >> INVITE to the new contacts? if so, take a look at the uac_redirect module. >> >> regards, >> bogdan >> >> Shane Burrell wrote: >> >> >>> I have this working to the point it tries to relay the call but I see >>> that internally openser is not making the new call. I assume t_relay >>> calls a new route but how to I make sure that all relays get >>> forwarded. Some of my logic else where seems to be killing the >>> rerouted call and I need any 302 routed calls to be accepted and routed. >>> >>> >>> >>> TIA, >>> >>> >>> >>> Shane >>> >>> >>> *************************************************************** >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > > >

1 0

[Users] TCP lifetime
by Klaus Darilion 03 Apr '07

03 Apr '07

Hi! I thought setting the tcp_connection_lifetime to 0 should keep them open endlessly, but in my case the proxy closes the TCP connection ~10 seconds after the last SIP message. What am I doing wrong? regards klaus -- Klaus Darilion nic.at

2 7

[Users] NEW FEATURE: IP blacklists
by Bogdan-Andrei Iancu 03 Apr '07

03 Apr '07

Hi everybody, OpenSER 1.2.0 has new feature - IP Blacklist support. This is a low level filtering engine for the outgoing requests; low level, because the filtering is done based on IP, protocol, port, etc. Its primary purposes will be to prevent sending requests to critical IPs (like GWs) due DNS or to avoid sending to destinations that are known to be unavailable (temporary or permanent). Because of flexibility concerns, the filtering rules can be groups inside multiple lists. A rule: - matches based on IP/mask, proto, port and text pattern criteria - can be reversed applied A list: - can be read-only - it does not change during execution - have timeout per elements - elements expires after a configured timeout. How to use: =========== currently there are 2 ways of using the blacklists: 1) statically defining list in the configuration file and selecting which ones should be used for each request. You can define blacklists as follow: # filter out requests going to ips of my gws dst_blacklist = gw:{( tcp , 192.168.2.100 , 5060 , "" ),( any , 192.168.2.101 , 0 , "" )} # block requests going to "evil" networks dst_blacklist = net_filter:{ ( any , 192.168.1.100/255.255.255.0 , 0 , "" )} # block message requests with nasty words dst_blacklist = msg_filter:{ ( any , 192.168.20.0/255.255.255.0 , 0 , "MESSAGE*ugly_word" )} # block requests not going to a specific subnet dst_blacklist = net_filter2:{ !( any , 192.168.30.0/255.255.255.0 , 0 , "" )} a rule is defined by: protocol : TCP, UDP, TLS or "any" for anything port : number or 0 for any ip/mask test patter - is a filename like matching (see "man 3 fnmatch") applied on the outgoing request buffer (first_line+hdrs+body) From routing script, you can use the use_blacklist("name") function to select what blacklist to be applied for the current request. More than one list can be selected. If the destination address matches on of the selected rules, the send will fail. 2) via DNS The DNS resolver, when configured with failover, can automatically store in a temporary blacklist the failed destinations. This will prevent (for a limited period of time) openser to send requests to destination known as failed. So, the blacklist can be used as a memory for the DNS resolver. To use it, you have to enabled it - the rest is done automatically. disable_dns_blacklist = no By default is enabled. The temporary blacklist created by DNS resolver is named "dns" and it is by default selected for usage (no need use the use_blacklist() function. The rules from this list have a life time of 4 minutes - you can change it at compile time, from blacklists.h . To give you an internal snapshot, a new MI function - "list_blacklists" - was added to print all existent blacklists and their rules. Any suggestions/reports are welcome! regards, bogdan

2 5

re: [Users] SIPp and registration
by yanlin 03 Apr '07

03 Apr '07

Hi, Scott Yagel, you could find some clues in the file scenario.cpp of sipp project. there were some embeded scenario scripts in sipp, stored in char* default_scenario[], you can sent a register sip message by doing this: " <send retrans=\"500\">\n" " <![CDATA[\n" "\n" " REGISTER sip:CA.cym.com SIP/2.0\n" " Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]\n" " From: ua1 <sip:ua1@nnl.cym:[local_port]>;tag=[pid]SIPpTag07[call_number]\n" " To: ua1 <sip:ua1@nnl.cym:[local_port]>\n" " Call-ID: [call_id]\n" " CSeq: 1 REGISTER\n" " Contact: sip:ua1@[local_ip]:[local_port]\n" " Content-Length: 0\n" " Expires: 300\n" "\n" " ]]>\n" " </send>\n" regards. Yan Lin

2 1

[Users] What does this mean?
by nick 03 Apr '07

03 Apr '07

pre_auth(): Credentials with given realm not found After looking around google it seems to be based on the fact that the first sip: value doesn't contain the userid (if I understand correctly...) I can't figure out exactly why the RURI is coming in this fashion.. Should I be changing something in the clients, or my config?? These worked on an older config, now with this section, they just don't get accepted. I've tried it with and without sipserver.mobilia.it in the realm. I've also tried it with the server generating ha1 and static ha1 (from the ha1 and ha1b fields in the db), no difference. I have use domain 1 in registrar and auth_db, and strip "sipserver." sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "1"); exit; } Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: New request - M=REGISTER RURI=sip:mobilia.it F=sip:rice@mobilia.it T=sip:rice@mobilia.it IP=x.x.x.105 ID=fmq-17240(a)x.x.x.105 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=100 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: get_hdr_field: cseq <CSeq>: <100> <REGISTER> Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=200 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: DEBUG: get_hdr_body : content_length=0 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: found end of header Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: find_first_route: No Route headers found Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: loose_route: There is no Route HF Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=ffffffffffffffff Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: check_via_address(x.x.x.105, x.x.x.105, 0) Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: parse_headers: flags=4000 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: pre_auth(): Credentials with given realm not found Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: Register authentication failed - M=REGISTER RURI=sip:mobilia.it F=sip:rice@mobilia.it T=sip:rice@mobilia.it IP=x.x.x.105 ID=fmq-17240(a)x.x.x.105 Mar 29 12:07:40 sipserver /usr/local/sbin/openser[22732]: build_auth_hf(): 'WWW-Authenticate: Digest realm="mobilia.it", nonce="460b9118a6a044d8837a01f924be2a9a3b7e0271", qop="auth"^M '

4 5

[Users] Simple xcap-emulator for OpenSER
by Adrian Georgescu 02 Apr '07

02 Apr '07

Hello, We wrote a simple xcap emulator to be able to manage the XCAP documents used by OpenSER presence agent, by using PHP scripts in combination with Apache web server. I have tested it against OpenSER with MySQL and Eyebeam for Apple. Klaus Darilion has tested it against Postgress and Eyebeam for Windows. The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-0.5.tar.gz Regards, Adrian

2 3

[Serusers] How to disable RTP proxy connection in nathelper module
by tzieleniewski 02 Apr '07

02 Apr '07

Hi, I am using nathelper and because I have an CISCO router in front of the SER I don't need to use RTP Proxy. Whenever I start SER I get the following errors: Apr 2 09:13:33 sen ser[3442]: ERROR: send_rtpp_command: can't connect to RTP proxy Apr 2 09:13:33 sen ser[3442]: send_rtpp_command(): proxy <unix:/var/run/rtpproxy.sock> does not responding, disable it Apr 2 09:13:33 sen ser[3442]: WARNING: rtpp_test: can't get version of the RTP proxy Apr 2 09:13:33 sen ser[3442]: WARNING: rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock>has been disabled temporarily Apr 2 09:13:33 sen ser[3444]: ERROR: send_rtpp_command: can't connect to RTP proxy Is there a way to disable this trials through ser.cfg by nathelper mod parameter in order not to get those errors? Cheers Tomasz

2 1

[Users] WeSIP C2C and freeswitch
by Helmut Kuper 02 Apr '07

02 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to use WeSIP C2C with UACs registered to freeswitch. freeswitch never answeres the INVITEs for caller (that INVITE with no SDP and from=to=caller's sip address). So my question is, is it because freeswitch maybe doesn't support rfc 3725? If so, is there any C2C proven SIP-PBX I can use instead? Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7s8L4tZeNddg3dwRArGjAKCRJ+Kqi4FqdE2J9S/Q6t8L3VEVzwCgncCc G2rxVs9s3fqs+XgUXnsefRs= =RqW7 -----END PGP SIGNATURE-----

4 9

[Users] How URI is parsed...
by Stuart Sheldon 02 Apr '07

02 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Newbe question: Can someone explain what causes a match when lookup() and does_uri_exist() is used? Does it look at the least significant characters is the URI? Does it drop certain leading numbers like '1'? Stu -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQFGDs0BK69Y+xPZrWYRAg5XAJ9kL3FNuImfu6wasPZ9z32uDYmoGwCfWKhW 5al96b0dhkmFNYnwBzyHmoY= =mNnn -----END PGP SIGNATURE-----

2 2

[Users] use of AVPs in core functions
by Jens Carl 02 Apr '07

02 Apr '07

Hello List, is it possible to use AVPs in core function like rewritehost(), etc. Regards Jens

2 1

Re: [Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Henning Westerholt 02 Apr '07

02 Apr '07

On Tuesday 27 March 2007 17:41, you wrote: > Hi Henning. > > I tested the compilation and it does not detect the architecture. > > I think that maybe is because of uname -p outputs sparc, and uname -m > outputs sun4v. > [..] Hi Sergio, thank you for the testing. I add another check to the patch according your suggestions, could please try it another time? Cheers, Henning

3 2

re: [Users] postgres
by Mark Price 02 Apr '07

02 Apr '07

On 12/20/06, Klaus Darilion <klaus.mailinglists(a)pernau.at > wrote: > > Hi Mark! > > Postgres should work well - I use it since ser 0.8. Just make sure that > the hard disk does not get full, because this breaks the index inside > postgres and postgres is getting real slow (re-create the index if it > happens). > > Last time I tested openser_postgres.sh it worked fine. If you find a bug > please let us know. > > Please check to use the latest versions from CVS (for 1.1 use CVS > rel_1_1_0) The latest version from cvs doesn't work out of the box with openser build from the release tarball, because the release tarball doesn't included /usr/sbin/openser_gen_ha1 (although I could get past this by replacing it with the md5sum equivelant). However, the release tarball doesn't work because of this bug: https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_… So, 1. is the cvs version of rel_1_1_0 considered safe for production use? 2. if so and there are important fixes such as this in CVS, would openser.org consider releasing a point release containing such changes? Thanks, Mark Price

6 8

RE: [Users] out of memory
by Papadopoulos Georgios 02 Apr '07

02 Apr '07

Hi Bogdan, I increased PKG_MEM_POOL_SIZE to 2MB and of course it took much longer to run out of memory. Usually it happens once every day, this time it ran for 3 days before running out of memory. If you don't mind, I will send the memory dump by personal email since I don't have easy access to an ftp server. Can you explain what you mean by 1.1.x? The version I am using is openser-1.1.1-notls_src.tar.gz, downloaded from your site under 1.1.1/. I am guessing that to get the fix I should use whatever is under latest-1.1.x/ Is this right? thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, March 30, 2007 3:33 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory > > Papadopoulos Georgios wrote: > > > > Hi Bogdan, > > > > I found the following thread in the list > > http://www.openser.org/pipermail/users/2006-June/005339.html > > I suspect that I have the same issue. If I assume that our DNS is > > strange, then my 1.1.1-notls version will be affected, > right? Is there > > a patch to fix this? Eventually I want to go to 1.2 but I > have to do a > > lot of testing before I can put it in production. So for > the moment I > > need a fix for 1.1.1. > > > that fix is present in 1.1.x also. > > Also back in December I sent this to the list > > http://www.openser.org/pipermail/users/2006-December/008205.html > > >From the start, we had the issue of openser stopping to process any > > incoming requests. On December I thought it was DNS related. I am > > still in the dark about this. Lately I added logging and I > discovered > > the "out of memory" error. I am guessing that the error was > there from > > the start, I just did not know about it. Do all of these make any > > sense to you? And is there a way to test our DNS to > determine whether it is buggy? > > > I;m afraid not :) > > In the meantime I will follow the instructions on the wiki > to get the > > memory dump. > > > yes - that will be the most relevant info. > > regards, > bogdan > > thank you > > > > George > > > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Wednesday, March 28, 2007 11:54 AM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] out of memory > >> > >> Hi Papadopoulos, > >> > >> the error you get complains about the lack of private memory (per > >> process) - the statistics show info about the shared memory. See: > >> http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > >> > >> but look for pkg instead of shm..... > >> > >> regards, > >> bogdan > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello, > >>> > >>> Lately our OpenSER 1.1.1 is running out of memory and I get the > >>> following message > >>> 4(25190) get_hdr_field: out of memory The result is that > it stops > >>> processing any incoming request. This is a production > system and of > >>> course it causes a great deal of frustration. > >>> > >>> The last time this happened I got the following statistics before > >>> restarting > >>> > >>>> ./openserctl fifo get_statistics > >>>> > >>> Total statistics = 37 > >>> Total modules = 5 > >>> Module name = core; statistics=8 > >>> core:rcv_requests = 3259334 > >>> core:rcv_replies = 8816 > >>> core:fwd_requests = 736 > >>> core:fwd_replies = 137 > >>> core:drop_requests = 743 > >>> core:drop_replies = 0 > >>> core:err_requests = 0 > >>> core:err_replies = 43 > >>> Module name = shmem; statistics=6 > >>> shmem:total_size = 268435456 > >>> shmem:used_size = 1675864 > >>> shmem:real_used_size = 1835104 > >>> shmem:max_used_size = 4399048 > >>> shmem:free_size = 266600352 > >>> shmem:fragments = 17710 > >>> Module name = sl; statistics=9 > >>> sl:1xx_replies = 0 > >>> sl:2xx_replies = 1690410 > >>> sl:3xx_replies = 43 > >>> sl:4xx_replies = 334827 > >>> sl:5xx_replies = 1185629 > >>> sl:6xx_replies = 0 > >>> sl:sent_replies = 3210909 > >>> sl:sent_err_replies = 0 > >>> sl:received_ACKs = 743 > >>> Module name = tm; statistics=11 > >>> tm:received_replies = 8636 > >>> tm:relayed_replies = 7096 > >>> tm:local_replies = 1718 > >>> tm:UAS_transactions = 7003 > >>> tm:UAC_transactions = 94 > >>> tm:2xx_transactions = 1528 > >>> tm:3xx_transactions = 3 > >>> tm:4xx_transactions = 5129 > >>> tm:5xx_transactions = 468 > >>> tm:6xx_transactions = 0 > >>> tm:inuse_transactions = 0 > >>> Module name = usrloc; statistics=3 > >>> usrloc:location-users = 0 > >>> usrloc:location-contacts = 0 > >>> usrloc:location-expires = 7282 > >>> > >>> I am not sure how to read these numbers but I see a big > "free_size" > >>> and still OpenSER is out of memory. Any suggestions? > >>> > >>> 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. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >> > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > > >

3 2

[Users] OpenSER v1.2.0 released
by Daniel-Constantin Mierla 02 Apr '07

02 Apr '07

Hello, version 1.2.0 of OpenSER has been released. It is major release, which means that you get along a lot of new features and improvements to existing ones in old versions. The configuration file and database structure suffered some changers, please read careful release notes to learn how to update. Just to list few of the new features: - SIMPLE Presence support (http://openser.org/dokuwiki/doku.php/presence:presence-module) - SNMP (http://openser.org/docs/modules/devel/snmpstats.html) - PERL API (http://openser.org/docs/modules/devel/perl.html) - JAVA SIP Servlet Application Server (http://wesip.eu) - script variables and transformations (http://openser.org/dokuwiki/doku.php/transformations:devel) - DNS failover - IP blacklists - XMPP IM gateway - XMLRPC - SIP session timers - secure federation peering - retransmission timer accuracy - usrloc refurbished for better performances - automatic error handling - AVPs in reply routes Read full release notes at: http://www.openser.org/mos/view/OpenSER-v1.2.x-Release-Notes/ Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or SourceForge: http://sourceforge.net/project/showfiles.php?group_id=139143&package_id=224… This is the first release coming out o SVN. Old CVS repository was migrated to SVN, and with these, a few changes apply to get sources from repository. Please read instructions from download page: http://www.openser.org/mos/view/Download/ Migration instructions (although not complete, I'm sure) are available at: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x The documentation for modules is posted at: http://openser.org/docs/modules/1.2.x/ We would like to thank to all developers and contributors for their work with coding and testing for this new release. Cheers, Daniel

9 20

[Serusers] Header Selects
by Andrey Kuprianov 02 Apr '07

02 Apr '07

Hi, @msg.event or @msg.supported, or even @msg.["header-name"] is not implemented yet? im using latest ser from CVS.. btw, is it ok to use latest SER 2.0 for presence instead of the suggested snapshot v.4? I've noticed xcap-root and auth_xcap_root params were removed from rls and pa modules, so i've set xcap_root param in xcap module instead.

3 6

[Users] presence logging....
by Helge Waastad 02 Apr '07

02 Apr '07

Hi, I'm just wondering if it's possible to turn off logging of some presence status messages? I have debug=0 But still I see a lot of: pr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: generate_ETag: etag= a.1175493921.28662.1 / 20 Apr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: get_subs_dialog:The query for subscribtion for [user]= xxx,[domain]= yyy for [event]= presence returned no result Apr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE:query_db_notify: Could not get subs_dialog from database /32] 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: handle_publish: sip-etag: [SIP-ETag: a.1175493921.28662.1 Is the only way do only have L_ERR filtered in syslog daemon? br hw

1 0

[Serusers] About avp_db_store()
by chungyu 02 Apr '07

02 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("$ruri/username", "s:name"); avp_db_store("$ruri", "s:name/contact"); }; if the incoming call then save username and domain of caller in contact caller: bob(a)iptel.org => ----------------------------- | username | domain | ----------------------------- | bob | iptel.org | ----------------------------- | | | ----------------------------- but not success, system reply ERROR:avpops:db_store: insert faild Please help me in the problem Thanks a lot in advance chungyu

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