- sr-users - kamailio.org

[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 >> > >> >

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[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....

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[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..."

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[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

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

23 Apr '07

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

9 10

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

23 Apr '07

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

1 0

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

23 Apr '07

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

1 0

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

23 Apr '07

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

9 25

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

23 Apr '07

Hello all, In the following call to PSTN, Openser gets into an infinite loop when the ACK is received. It starts sending the ACK to itself. Is this because the ACK is not right or could it be a misconfiguration in my part? thank you for any help George # U 2007/04/18 13:44:46.759951 213.5.17.223:5058 -> 213.5.43.4:5060 INVITE sip:2116873831@sip.altecnet.gr:5060 SIP/2.0. Max-Forwards: 70. Content-Length: 378. Via: SIP/2.0/UDP 213.5.17.223:5058;branch=z9hG4bK08349b43d. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>. CSeq: 424044974 INVITE. Supported: 100rel. Supported: timer. Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE. Content-Type: application/sdp. Supported: replaces. Proxy-Authorization:Digest response="87bc0f48f292fa5213368434f8712c39",username="demo1",realm="alte cnet.gr",nonce="4625f7cabd05f0720edd9ab601211b1759066235",uri="sip:21168 73831@sip.altecnet.gr:5060". Contact: demo1 <sip:demo1@213.5.17.223:5058>. User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. . v=0. o=MxSIP 0 1153034350 IN IP4 213.5.17.223. s=SIP Call. c=IN IP4 213.5.17.223. t=0 0. m=audio 57994 RTP/AVP 8 0 97 98 109 99 4 18. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:97 G726-16/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:109 G726-32/8000. a=rtpmap:99 G726-40/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729/8000. a=ptime:20. a=sendrecv. a=silenceSupp:on - - - -. # U 2007/04/18 13:44:46.764816 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 213.5.17.223:5058;branch=z9hG4bK08349b43d;rport=5058. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>. CSeq: 424044974 INVITE. Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. . # U 2007/04/18 13:44:46.764967 213.5.43.4:5060 -> 213.5.1.6:5060 INVITE sip:400#2116873831@213.5.1.6:5060 SIP/2.0. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Max-Forwards: 69. Content-Length: 378. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>. CSeq: 424044974 INVITE. Supported: 100rel. Supported: timer. Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE. Content-Type: application/sdp. Supported: replaces. Contact: demo1 <sip:demo1@213.5.17.223:5058>. User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. . v=0. o=MxSIP 0 1153034350 IN IP4 213.5.17.223. s=SIP Call. c=IN IP4 213.5.17.223. t=0 0. m=audio 57994 RTP/AVP 8 0 97 98 109 99 4 18. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:97 G726-16/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:109 G726-32/8000. a=rtpmap:99 G726-40/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729/8000. a=ptime:20. a=sendrecv. a=silenceSupp:on - - - -. # U 2007/04/18 13:44:46.781896 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2007/04/18 13:44:46.913667 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:46.914048 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:47.413787 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:47.414278 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.413460 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.413952 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Require: 100rel. RSeq: 1883. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.582989 213.5.1.6:5060 -> 213.5.43.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.0,SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.583304 213.5.43.4:5060 -> 213.5.17.223:5058 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bK08349b43d. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Date: Wed, 18 Apr 2007 10:44:46 GMT. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 424044974 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116873831@213.5.1.6:5060>. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Content-Type: application/sdp. Content-Length: 232. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7897 3551 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 19140 RTP/AVP 18. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:40. a=silenceSupp:off - - - -. # U 2007/04/18 13:44:48.718798 213.5.17.223:5058 -> 213.5.43.4:5060 ACK sip:2116873831@sip.altecnet.gr:5060 SIP/2.0. Max-Forwards: 70. Content-Length: 0. Via: SIP/2.0/UDP 213.5.17.223:5058;branch=z9hG4bKd1e7e81f7. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. CSeq: 424044974 ACK. Proxy-Authorization:Digest response="87bc0f48f292fa5213368434f8712c39",username="demo1",realm="alte cnet.gr",nonce="4625f7cabd05f0720edd9ab601211b1759066235",uri="sip:21168 73831@sip.altecnet.gr:5060". User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. . # U 2007/04/18 13:44:48.719303 213.5.43.4:5060 -> 213.5.43.4:5060 ACK sip:213.5.43.4;lr=on;ftag=299ad500d129e32 SIP/2.0. Record-Route: <sip:213.5.43.4;lr=on;ftag=299ad500d129e32>. Max-Forwards: 69. Content-Length: 0. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bKb455.b4c98796.2. Via: SIP/2.0/UDP 213.5.17.223:5058;rport=5058;branch=z9hG4bKd1e7e81f7. Call-ID: 91a3a06767e6d11c4a62a583078d95fb(a)213.5.17.223. From: demo1 <sip:demo1@sip.altecnet.gr:5060>;tag=299ad500d129e32. To: 2116873831 <sip:2116873831@sip.altecnet.gr:5060>;tag=4C655430-F57. CSeq: 424044974 ACK. Proxy-Authorization:Digest response="87bc0f48f292fa5213368434f8712c39",username="demo1",realm="alte cnet.gr",nonce="4625f7cabd05f0720edd9ab601211b1759066235",uri="sip:21168 73831@sip.altecnet.gr:5060". User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5. P-hint: rr-enforced. . Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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Re: [Users] Re: Multiple invites
by venkatesh.d 23 Apr '07

23 Apr '07

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

1 0

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

23 Apr '07

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

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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

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[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

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[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

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[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

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[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

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[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.

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[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.

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

20 Apr '07

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

1 0

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

20 Apr '07

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

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[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 >

2 2

[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

1 0

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(); > }

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

20 Apr '07

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

5 6

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

20 Apr '07

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

3 2

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

20 Apr '07

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

2 1

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

20 Apr '07

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

2 1

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

19 Apr '07

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

3 2

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

19 Apr '07

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

2 1

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

19 Apr '07

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

2 2

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

19 Apr '07

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

2 1

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

19 Apr '07

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

2 1

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

19 Apr '07

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

2 2

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

19 Apr '07

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

1 0

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

19 Apr '07

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

2 1

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

19 Apr '07

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

2 8

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

19 Apr '07

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

2 1

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

19 Apr '07

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

3 2

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

19 Apr '07

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

2 3

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

19 Apr '07

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

4 9

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

19 Apr '07

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

2 1

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

19 Apr '07

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

2 1

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

19 Apr '07

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

2 1

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

18 Apr '07

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

2 1

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

18 Apr '07

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

2 7

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

18 Apr '07

Dear all, i am trying forwarding call with openser .i am using mediaproxy with openser. call is not getting forwarded uri.when i call UA1 - UA2 same call have to forwarded to UA3 but two INVITE messages are coming to UA2 . call is not going to UA3 .i am sending openser.cfg file also . please help me on this. at line 190 in openser.cfg am trying to forward using append_branch() . Thanks and Regards venkatesh.d debug=6 fork=yes log_stderror=yes listen=192.168.0.248 port=5060 children=4 #dns=no #rev_dns=no #debug=9 # debug level (cmd line: -ddddddddd #fork=yes #log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes #check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) #port=5060 #children=4 #fifo="/tmp/openser_fifo" fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openser@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/presence.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #modparam("rr", "enable_full_lr", 1) modparam("presence", "db_url","mysql://openser:openser@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') modparam("presence", "totag_avpid", 987) modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "force_active", 1) #modparam("avpops", "avp_aliases", "uuid=I:50") route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("sip.pyrogroup.com:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="ACK") { route(1); exit; } else if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); ###Call Forwaring here append_branch("sip:shekhar@58.68.74.84"); if (!t_relay()) { # if (method="INVITE" || method=="ACK") { # end_media_session(); # }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("localhost","subscriber")) { www_challenge("localhost","0"); #consume_credentials(); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

1 0

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

18 Apr '07

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

2 4

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

18 Apr '07

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

2 1

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

18 Apr '07

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

2 1

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

18 Apr '07

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

2 1

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

18 Apr '07

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

2 1

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

18 Apr '07

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

2 2

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

18 Apr '07

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

3 4

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

17 Apr '07

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

3 4

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

17 Apr '07

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

2 1

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

17 Apr '07

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

2 1

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

17 Apr '07

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

1 0

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

17 Apr '07

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

3 3

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

17 Apr '07

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

2 1

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

17 Apr '07

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

2 3

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

16 Apr '07

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

2 2

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

16 Apr '07

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

1 0

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

16 Apr '07

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

1 0

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

16 Apr '07

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

3 10

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

16 Apr '07

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

3 6

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

16 Apr '07

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

2 1

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

16 Apr '07

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

2 1

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

15 Apr '07

Hi with openser 1.1.1 I have this error Apr 15 10:27:18 byll4 openser[3844]: ERROR: tsend_datagram_ev: failed to send: (90) Message too long Apr 15 10:27:18 byll4 openser[3844]: ERROR:tm:write_to_unixsock: writev failed: Message too long Apr 15 10:27:18 byll4 openser[3844]: ERROR:tm:t_write_unix: write_to_unixsock failed Laurent # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes # yes log_stderror=no #yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 unix_sock="/tmp/ser_sock" listen=udp:195.129.125.57:5060 # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/rr/rr.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/textops/textops.so" #loadmodule "modules/avp/avp.so" loadmodule "modules/avpops/avpops.so" loadmodule "modules/xlog/xlog.so" #loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- modparam("xlog", "buf_size", 8192) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam( "avpops", "avp_aliases", "billing_data=i:67" ) modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append", "billing_headers: ua=$hdr(User-Agent)") #modparam("tm", "tw_append", "billing_headers: ua=$hdr(User-Agent) ;pbilling=$avp(i:67);x-billingId=$hdr(x-billingId);remote-party-id=$hdr(Remote-Party-ID)") modparam("tm", "fr_timer", 200) # appends for dtmf per INFO #modparam( "tm", "tw_append", # "info_append: ctlength=$hdr(Content-Length); cttype=$hdr(Content-Type); msg=$msg(body)") # main routing logic route{ xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); return; } sl_send_reply("100", " trying "); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); log("step 1\n"); xlog("L_ERR", "step 1\n"); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { log("loose route\n"); xlog("L_ERR", "loose route\n"); append_hf("P-hint: rr-enforced\r\n"); if (!t_relay()) { sl_reply_error(); } return; }; xlog("L_ERR", "step t_newtran\n"); # make transaction # if (method != "ACK") { if (!t_newtran()){ log("could not create transaction\n"); xlog("L_ERR", "could not create transaction \n"); sl_send_reply("500","could not create transaction"); return; } #} log("SIP message \n"); xlog("L_ERR", "step 2\n"); # actively absorb ACKs if (method == "ACK") { log("ACK ... \n"); xlog("L_ERR", "ACK ...\n"); if (!t_relay()) { sl_reply_error(); } #t_relay(); # break; } # pass INFO to SEMS # if (method=="INFO") { # if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ # log("could not contact sems\n"); # t_reply("500","could not contact media server"); # } # } if (uri =~ "sip:.*@") { xlog("L_ERR", "to sems\n"); # if (method=="INVITE") # { # avp_write("$hdr[x-billingId]","$billing_data"); #} if (!t_write_unix("/tmp/sems_sock","billing/billing_headers")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } t_reply("404","Not found in ser .... 1");

2 1

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

15 Apr '07

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

2 1

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

14 Apr '07

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

4 5

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

14 Apr '07

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

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

14 Apr '07

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

2 1

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

13 Apr '07

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

2 4

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

13 Apr '07

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

1 0

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

13 Apr '07

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

2 6

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

13 Apr '07

Dear friends: I have installed openser-1.1.0-tls version, and it works fine with mysql, the sip flow is below: UA OpenSER invite----------------> <----------------100 trying <----------------101 dialog establishement <----------------180 ringing ..... My question is, why not have the 407 Proxy Authentication Required in the per call, see my openser.cfg file: # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } ============================ thanks. Mike --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

3 2

[Serusers] using SER
by gabrielyans 13 Apr '07

13 Apr '07

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

3 2

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

13 Apr '07

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

1 0

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

13 Apr '07

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

3 3

[Serusers] fifo
by Javier 12 Apr '07

12 Apr '07

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

2 1

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

12 Apr '07

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

3 2

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

12 Apr '07

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

2 1

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

12 Apr '07

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

3 4

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

12 Apr '07

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

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[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

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[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....

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