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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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