sr-users May 2005

sr-users@lists.kamailio.org

230 participants
522 discussions

Re: [Serusers] Error loading mysql module - undefined symbol fm_malloc

by a c

and # make prefix=/usr/local install Makefile.defs:460: Old gcc detected (2.9x), use gcc >= 3.1 for better results --- a c <lra101(a)yahoo.com> wrote: > I don't see any errors but could the gcc version > make > a difference? > > %gcc -v > Reading specs from > /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs > gcc version 2.96 20000731 (Red Hat Linux 7.1 > 2.96-81) > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > I would check one more time and make sure that > > compilation and linking give > > no errors/warnings... > > g-) > > > > a c wrote: > > > hello Greger, > > > > > > seems like it is already commented out..hmm > > > > > > > > > DEFS+= $(extra_defs) \ > > > -DNAME='"$(MAIN_NAME)"' > > > -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' \ > > > -DOS='"$(OS)"' -DCOMPILER='"$(CC_VER)"' > > > -D__CPU_$(ARCH) -D__OS_$(OS) \ > > > -DCFG_DIR='"$(cfg-target)"'\ > > > -DPKG_MALLOC \ > > > -DSHM_MEM -DSHM_MMAP \ > > > -DDNS_IP_HACK \ > > > -DUSE_IPV6 \ > > > -DUSE_MCAST \ > > > -DUSE_TCP \ > > > -DDISABLE_NAGLE \ > > > #-DDBG_QM_MALLOC \ > > > #-DF_MALLOC \ > > > #-DDBG_F_MALLOC \ > > > #-DNO_DEBUG \ > > > #-DNO_LOG \ > > > #-DVQ_MALLOC \ > > > #-DCONTACT_BUG \ > > > #-DDBG_LOCK \ > > > #-DNOSMP \ > > > #-DEXTRA_DEBUG \ > > > #-DUSE_SHM_MEM \ > > > #-DSTATS \ > > > #-DNO_LOG > > > > > > > > > --- "Greger V. Teigre" <greger(a)teigre.com> > wrote: > > > > > >> Looks like you have compiled with a memory > alloc > > >> function not available. > > >> Check your Makefile.defs file. There are many > > >> different options. Look at > > >> this section: > > >> DEFS+= $(extra_defs) \ > > >> -DNAME='"$(MAIN_NAME)"' > > >> -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' > > >> \ > > >> -DOS='"$(OS)"' > -DCOMPILER='"$(CC_VER)"' > > >> -D__CPU_$(ARCH) -D__OS_$(OS) > > >> \ > > >> -DCFG_DIR='"$(cfg-target)"'\ > > >> -DPKG_MALLOC \ > > >> -DSHM_MEM -DSHM_MMAP \ > > >> -DDNS_IP_HACK \ > > >> -DUSE_IPV6 \ > > >> -DUSE_MCAST \ > > >> -DUSE_TCP \ > > >> -DDISABLE_NAGLE \ > > >> -DDBG_QM_MALLOC \ > > >> -DF_MALLOC \ > > >> #-DDBG_F_MALLOC \ > > >> #-DNO_DEBUG \ > > >> #-DNO_LOG \ > > >> #-DVQ_MALLOC \ > > >> #-DCONTACT_BUG \ > > >> #-DDBG_LOCK \ > > >> #-DNOSMP \ > > >> #-DEXTRA_DEBUG \ > > >> #-DUSE_SHM_MEM \ > > >> #-DSTATS \ > > >> #-DNO_LOG > > >> > > >> You can comment out F_MALLOC > > >> g-) > > >> > > >> a c wrote: > > >>> hello, > > >>> > > >>> having some problems loading the mysql module. > I > > am > > >>> using the latest dev release. couldn't find > much > > help > > >>> in the archieves.. > > >>> > > >>> ser -cf ser.mysql.cfg > > >>> 0(1439) ERROR: load_module: could not open > > module > > >>> </usr/local/lib/ser/modules/mysql.so>: > > >>> /usr/local/lib/ser/modules/mysql.so: undefined > > symbol: > > >>> fm_malloc > > >>> 0(1439) parse error (15,13-14): failed to load > > module > > >>> ERROR: bad config file (1 errors) > > >>> > > >>> > > >>> ----------------config------------------ > > >>> debug=3 > > >>> fork=yes > > >>> log_stderror=no > > >>> > > >>> listen=192.168.0.5 > > >>> port=5060 > > >>> children=4 > > >>> > > >>> dns=no > > >>> rev_dns=no > > >>> > > >>> fifo="/tmp/ser_fifo" > > >>> fifo_db_url="mysql://ser:heslo@localhost/ser" > > >>> > > >>> loadmodule > "/usr/local/lib/ser/modules/mysql.so" > > >>> > > >>> > > >>> > > __________________________________________________ > > >>> Do You Yahoo!? > > >>> Tired of spam? Yahoo! Mail has the best spam > > protection around > > >>> http://mail.yahoo.com > > >>> > > >>> > _______________________________________________ > > >>> Serusers mailing list > > >>> serusers(a)lists.iptel.org > > >>> > http://lists.iptel.org/mailman/listinfo/serusers > > >> > > >> > > > > > > > > > > > > > > > __________________________________ > > > Do you Yahoo!? > > > Yahoo! Small Business - Try our new Resources > site > > > http://smallbusiness.yahoo.com/resources/ > > > > > > > > > __________________________________ > Do you Yahoo!? > Yahoo! Small Business - Try our new Resources site > http://smallbusiness.yahoo.com/resources/ > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 11 months

Re: [Serusers] Re: SER CANCEL Processing

by a c

thanks guys..t_relay was the problem. I did not have this in the config. issue is resolved. ac --- a c <lra101(a)yahoo.com> wrote: > I believe the 487 does not get matched because the > UA > which does the CANCEL never gets a 200 OK back since > the CANCEL was not send out to the terminating UA by > SER. For CANCEL, are we supposed to have special > config? I don't have any specifically handles CANCEL > {other than 487}. > > ac > > > --- Steve Blair <blairs(a)isc.upenn.edu> wrote: > > > > > I am having a similar problem. I am not yet > > convinced that it is SER > > as much as my configuration but if SER does have > > difficulty processing > > CANCEL messages it would be helpful to understand > > under what > > circumstances. > > > > By the way I have been using t_check_status in > a > > similar way since > > v 0.8 code. When I upgraded to 0.9.0, .0 and now > .2 > > the t_check_status > > fails to be matched. > > > > -Steve > > > > Thanks,Steve > > > > a c wrote: > > > > >any ideas anyone? seems like SER does not respond > > to > > >CANCEL requests and eventually takes the failure > > >route. > > > > > >I have the following in my failure route, but it > > never > > >gets hit, since SER does not send the 200 OK back > > or > > >pass the CANCEL message forward. log file > attached. > > > > > >failure_route[1] > > >{ > > > if(t_check_status("487")) { > > > # don't continue on cancellation > > > xlog("L_WARN", "OH OH OH\n"); > > > break; > > > } > > > > > >ac > > > > > > > > >--- a c <lra101(a)yahoo.com> wrote: > > > > > > > > > > > >>Seems like SER does not respond back to CANCEL > > >>Requests. Is this a issue / some config problem > on > > >>my > > >>side? ngrep traces attached > > >> > > >>interface: eth0 (192.168.0.0/255.255.255.0) > > >>filter: ip and ( port 5060 ) > > >># > > >>U 192.168.0.2:5062 -> 192.168.0.5:5060 > > >> INVITE sip:9999999999@voip.com SIP/2.0..Via: > > >>SIP/2.0/UDP 192.168.0.2:5062;r > > >> > > >> > > >> > > >> > > > >port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > > > > > > > > >>102 <sip:102@voi > > >> p.com:5062>;tag=3282942385..To: > > >><sip:9999999999@voip.com>..Contact: <sip:10 > > >> 2@192.168.0.2:5062>..Call-ID: > > >>29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. > > >> 0.2..CSeq: 940 INVITE..Max-Forwards: > > >>70..Content-Type: application/sdp..Use > > >> r-Agent: X-Lite release 1103m..Content-Length: > > >>258....v=0..o=102 7659613 76 > > >> 59633 IN IP4 192.168.0.2..s=X-Lite..c=IN IP4 > > >>192.168.0.2..t=0 0..m=audio 80 > > >> 00 RTP/AVP 0 8 3 98 96..a=rtpmap:0 > > >>pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpma > > >> p:3 gsm/8000..a=rtpmap:98 > iLBC/8000..a=rtpmap:96 > > >>telephone-event/8000..a=fm > > >> tp:96 0-15.. > > > > > >> > > >> > > >># > > >>U 192.168.0.5:5060 -> 192.168.0.2:5062 > > >> SIP/2.0 100 trying -- your call is important > to > > >>us..Via: SIP/2.0/UDP 192.16 > > >> > > >> > > >> > > >> > > > >8.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > > > > > > > > >> 102 > <sip:102@voip.com:5062>;tag=3282942385..To: > > >><sip:9999999999@voip.com>. > > >> .Call-ID: > > >> > > >> > > >> > > > >29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: > > > > > > > > >>940 INVIT > > >> E..Server: Sip EXpress router (0.9.2 > > >>(i386/linux))..Content-Length: 0..Warn > > >> ing: 392 192.168.0.5:5060 "Noisy feedback > tells: > > > > >>pid=2943 req_src_ip=192.1 > > >> 68.0.2 req_src_port=5062 > > >>in_uri=sip:9999999999@voip.com > out_uri=sip:9999999 > > >> 999(a)voip.com via_cnt==1".... > > > > > >> > > >> > > >># > > >>U 192.168.0.5:5060 -> 192.168.0.3:5060 > > >> INVITE sip:9999999999@voip.com > > >>SIP/2.0..Record-Route: <sip:192.168.0.5;ftag > > >> =3282942385;lr=on>..Via: SIP/2.0/UDP > > >>192.168.0.5;branch=z9hG4bKe034.1018c5d > > >> 1.0..Via: SIP/2.0/UDP > > >> > > >> > > >> > > > >192.168.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8C > > > > > > > > >> F4A479B6D9DDB17C5CFC1..From: 102 > > >><sip:102@voip.com:5062>;tag=3282942385..To > > >> : <sip:9999999999@voip.com>..Contact: > > >><sip:102@192.168.0.2:5062>..Call-ID: > > >> > > >> > > >> > > >> > > > >29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: > > > > > > > > >>940 INVITE..Max-For > > >> wards: 16..Content-Type: > > >>application/sdp..User-Agent: X-Lite release > 1103m. > > >> .Content-Length: 258....v=0..o=102 7659613 > > 7659633 > > >>IN IP4 192.168.0.2..s=X- > > >> Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio > 8000 > > >>RTP/AVP 0 8 3 98 96..a=rtpm > > >> ap:0 pcmu/8000..a=rtpmap:8 > pcma/8000..a=rtpmap:3 > > >>gsm/8000..a=rtpmap:98 iLBC > > >> /8000..a=rtpmap:96 > > telephone-event/8000..a=fmtp:96 > > >>0-15.. > > >># > > >>U 192.168.0.3:5060 -> 192.168.0.5:5060 > > >> SIP/2.0 100 Trying..From: 102 > > >><sip:102@voip.com:5062>;tag=3282942385..To: < > > >> sip:9999999999@voip.com>..Call-Id: > > >>29469D19-CDF1-49ED-8245-A67E1B1DBB88@192 > > >> .168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP > > >>192.168.0.5;branch=z9hG4bKe034 > > >> .1018c5d1.0..Via: SIP/2.0/UDP > > >>192.168.0.2:5062;rport=5062;branch=z9hG4bKE29 > > >> 503D9D8CF4A479B6D9DDB17C5CFC1..Content-Length: > > >>0.... > > >> > > >># > > >>U 192.168.0.3:5060 -> 192.168.0.5:5060 > > >> SIP/2.0 180 Ringing..From: 102 > === message truncated === __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

18 years, 11 months

Fw: [Serusers] Ser+Nat +Polycom phones

by Greger V. Teigre

Sorry, but for security reasons I don't read other people's dumps. You just need to check whether the m= line refers to your rtpproxy's public IP and not the private. And of course that rtpproxy and nathelper communicate with eachother... g-) > harry gaillac wrote: >> I've built rtpproxy with -i option : >> >> root 14362 1 0 16:39 ? 00:00:00 >> /usr/local/bin/rtpproxy -l 192.168.0.1 -i >> 80.119.11.112 >> >> However I can't send or receive invite from/to user >> agent . >> look at invitation file (ethereal)! >> Should I configure phone per port ? >> What's wrong ? >> >> Harry >> >> >> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit : >> >>> The patch was current for rtpproxy from repository >>> at that point. I sent the >>> patch to Maxim and asked if it could be included in >>> CVS, but I never heard >>> from him. >>> If you add the patch by hand, you should be able >>> to make rtpproxy >>> announce the public IP in the INVITEs and OKs >>> (instead of rtpproxy's >>> private/interface address). >>> g-) >>> >>> harry gaillac wrote: >>>> I tried to patch main.c without success according >>> to >>>> main.c.rej but i try to add -i by hand. >>>> >>>> I failed to patch main.c . >>>> >>>> If I add -i options to rtpproxy will it solve >>> problem >>>> of signaling behind nat (on same box) ? >>>> >>>> >>>> Harry >>>> >>>> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit >>>> >>>> >>>>> >>>> >>> >> http://lists.iptel.org/pipermail/serusers/2005-January/014688.html >>>>> harry gaillac wrote: >>>>>> Where can I get your patch ? >>>>>> >>>>>> Harry >>>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a >>> écrit >>>>>> >>>>>> >>>>>>> Running rtpproxy on a NATed box is a problem >>> (you >>>>>>> need a patch I posted a >>>>>>> while back to announce the correct public IP). >>>>>>> Running all daemons on the >>>>>>> same server with a public IP is NOT a problem. >>>>>>> g-) >>>>>>> harry gaillac wrote: >>>>>>>> Hello, >>>>>>>> >>>>>>>> SER+nathelper+rtpproxy run on the same box that nat. >>>>>>>> Is it a problem ? >>>>>>>> >>>>>>>> Harry >>>>>>>> >>>>>>>> >>>>>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a >>>>> écrit >>>>>>>> >>>>>>>> >>>>>>>>> That's strange. We are using some Polycom 300 >>>>> and >>>>>>>>> they work ok. No STUN >>>>>>>>> support and sometimes a very erratic >>>>> registration >>>>>>>>> pattern, but except from >>>>>>>>> that, they seem to perform ok. They >>>>> interoperate >>>>>>>>> with sipuras, >>>>>>>>> grandstreams, sjphone, cisco gw, xlite, etc. >>>>>>>>> g-) >>>>>>>>> >>>>>>>>> Juha Heinanen wrote: >>>>>>>>>> harry, >>>>>>>>>> >>>>>>>>>> we tried polycom video phones a month or so >>> ago and their sip >>>>>>>>>> implementation was worst i had seen so far. >>> the phones didn't >>>>>>>>>> support digest authentication and we were also hit the >>>>>>>>>> problem you mentioned with domain in request-uri. i suggest >>>>>>>>>> to junk those phones before their sip implementation is >>>>>>>>>> closer to >>> rfc3261. >>>>>>>>>> >>>>>>>>>> -- juha >>>>>>>>>> >>>>>>>>>> >>>>> _______________________________________________ >>>>>>>>>> Serusers mailing list >>>>>>>>>> serusers(a)lists.iptel.org >>>>>>>>>> >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>> >> _____________________________________________________________________________ >>>>>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage >>>>>>>> pour vos mails, photos et vidéos ! Créez votre Yahoo! >>> Mail sur >>>>>>>> http://fr.mail.yahoo.com >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>> >>> >> _____________________________________________________________________________ >>>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace >>>>> de stockage pour vos >>>>>> mails, photos et vidéos ! Créez votre Yahoo! Mail sur >>>>>> http://fr.mail.yahoo.com >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >> _____________________________________________________________________________ >>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace >>> de stockage pour vos >>>> mails, photos et vidéos ! Créez votre Yahoo! Mail >>> sur >>>> http://fr.mail.yahoo.com >>> >>> >> >> >> >> >> >> >> _____________________________________________________________________________ >> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos >> mails, photos et vidéos ! Créez votre Yahoo! Mail sur >> http://fr.mail.yahoo.com

18 years, 11 months

[Serusers] problems with NAT, ATA not responding to INVITEs

by Brian McCrary

Hello, I've been trying to add support for ATAs behind NAT firewalls for several days, and have had nothing but frustrations with it, so hopefully someone can clue me in. I tried modifying my existing ser config which has been working great, but failing that, I finally got the stock config file from ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.3.05.cfg, modified my IP address for NAT and used that (after adding the domain module). My first problem is that when an ATA registers behinf the NAT firewall, the NAT flag is never set, so if doesn't get the pings every 30 seconds. I can manually set that flag by just having setflag(6) in the config file and that seems to temporarily fix that problem, but I can't tell that it does anything else. Outbound calls from the ATA work just fine, which they always have even where I had no NAT stuff in SER at all. Inbound calls to the ATA get ignored my it though, this is a Cisco ATA-186. I hooked up an old hub so I could get the messages using ngrep on the side with the NAT router and the main server, and here's what I see. 7771111001 is the ATA, 7778881000 is a local phone. 11.11.11.100 is the SER router, 11.11.11.44 is the PSTN gateway initiating the call, 11.11.11.18 is the NAT router's public IP, and 10.0.2.3 is the ATA. SER router's ngrep: U 11.11.11.100:5060 -> 11.11.11.18:1387 27067@0:1480 ...k....INVITE sip:7771111001@11.11.11.18:1387;user=phone;transport=udp SI P/2.0..Record-Route: <sip:11.11.11.100;ftag=6B826304-1A6A;lr=on>..Via: SIP/2 .0/UDP 11.11.11.100;branch=z9hG4bKeda1.2545.3..Via: SIP/2.0/UDP 11.11.11.44 :5060;x-route-tag="tgrp:lnx";branch=z9hG4bK8C9D..From: <sip:7778881000@11.11 11.44>;tag=6B826304-1A6A..To: <sip:7771111001@11.11.11.100>..Date: Thu, 26 May 2005 14:10:42 GMT..Call-ID: C7AED903-CD2611D9-8130A2A3-EEBB2457(a)11.11. 11.44..Supported: 100rel,timer..Min-SE: 1800..Cisco-Guid: 3350084827-34418 24217-2183397385-1128595304..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 16..Remote-Party-ID : <sip:7778881000@11.11.11.44>;party=calling;screen=no;privacy=off..Timesta mp: 1117116642..Contact: <sip:7778881000@11.11.11.44:5060>..Expires: 180..A llow-Events: telephone-event..Content-Type: application/sdp..Content-Length : 506....v=0..o=CiscoSystemsSIP-GW-UserAgent 2462 1180 IN IP4 11.11.11.44.. s=SIP Call..c=IN IP4 11.11.11.44..t=0 0..m=audio 16536 RTP/AVP 18 2 0 100 1 01..c=IN IP4 11.11.11.44..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=yes..a=rt pmap:2 G726-32/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:100 X-NSE/8000..a=fmtp: 100 192-194,200-202..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a= X-sqn:0..a=X-cap: 1 audio RTP/AVP 100..a=X-cpar: a=rtpmap:100 X-NSE/8000..a =X-cpar: a=fmtp:100 192-194,200-202..a=X-cap: 2 image u NAT (ATA location) ngrep: U 11.11.11.100:5060 -> 10.0.2.3:5060 27070@0:1462 .......&INVITE sip:7771111001@11.11.11.18:1405;user=phone;transport=udp S IP/2.0..Record-Route: <sip:11.11.11.100;ftag=6B826304-1A6A;lr=on>..Via: SIP /2.0/UDP 11.11.11.100;branch=z9hG4bKeda1.2545.6..Via: SIP/2.0/UDP 11.11.11.44:5060;x-route-tag="tgrp:lnx";branch=z9hG4bK8C9D..From: <sip:7778881000@ 11.11.11.44>;tag=6B826304-1A6A..To: <sip:7771111001@11.11.11.100>..Date: Th u, 26 May 2005 14:10:42 GMT..Call-ID: C7AED903-CD2611D9-8130A2A3-EEBB2457@ 11.11.11.44..Supported: 100rel,timer..Min-SE: 1800..Cisco-Guid: 335008482 7-3441824217-2183397385-1128595304..User-Agent: Cisco-SIPGateway/IOS-12.x. .Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 16..Remot e-Party-ID: <sip:7778881000@11.11.11.44>;party=calling;screen=no;privacy=o ff..Timestamp: 1117116642..Contact: <sip:7778881000@11.11.11.44:5060>..Exp ires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..C ontent-Length: 478....v=0..o=CiscoSystemsSIP-GW-UserAgent 2462 1180 IN IP4 11.11.11.44..s=SIP Call..c=IN IP4 11.11.11.44..t=0 0..m=audio 16536 RTP/A VP 18 2 0 100 101..c=IN IP4 11.11.11.44..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=yes..a=rtpmap:2 G726-32/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:100 X- NSE/8000..a=fmtp:100 192-194,200-202..a=rtpmap:101 telephone-event/8000..a =fmtp:101 0-16..a=X-sqn:0..a=X-cap: 1 audio RTP/AVP 100..a=X-cpar: a=rtpma p:100 X-NSE/8000..a=X-cpar: a=fmtp:100 192-194,200-202.. It looks like the ATA is getting the message, but just flat out ignoring it. Anyone else have this problem? Brian

18 years, 11 months

RE: [Serusers] avp_radius and avpops

by Ricardo Martinez

Yes. I noticed that the variable used by the avp_load_radius("caller") to store the answer from the RADIUS server is in fact the one i defined first (var1), but is has a prefix called "caller_" , so the variable is finally : "caller_var1". Jackpot!. So i used this in my ser.cfg: if( !avp_load_radius("caller")) { xlog("L_INFO", "time [%Tf] FAIL AVP_LOAD_RADIUS!\n"); }; if( !avp_pushto("$ruri", "s:caller_var1/g")) { xlog("L_INFO", "time [%Tf] FAIL avp_pushto()!\n"); }; Thanks Greg for your help. Just one more thing. This could be indicated in the module's README, i think is an important thing that a user should know. Regards!!. Ricardo Martinez.- > -----Mensaje original----- > De: Greger V. Teigre [mailto:greger@teigre.com] > Enviado el: Viernes, 27 de Mayo de 2005 3:57 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: Re: [Serusers] avp_radius and avpops > > > Do the debugging dance, so that you can verify what the calls > to avp_* do... > g-) > > Ricardo Martinez wrote: > > Hello Greg > > Thanks for your answer. > > I made the change but it seems not to do anything.... > > Can someone help me here? > > > > Thanks in advance > > > > Regards > > Ricardo.- > > > >> -----Mensaje original----- > >> De: Greger V. Teigre [mailto:greger@teigre.com] > >> Enviado el: Miércoles, 25 de Mayo de 2005 16:24 > >> Para: Ricardo Martinez; serusers(a)lists.iptel.org > >> Asunto: Re: [Serusers] avp_radius and avpops > >> > >> > >>> if( !avp_pushto("$RURI", "s:var1/g")) { > >> replace with: > >> "s:$var1/g" > >> I don't know about the case sensivtivity of destination, I > >> always use lower > >> case... > >> g-) > >> > >> Ricardo Martinez wrote: > >>> Hello List. > >>> I have a question regarding the use of avp_radius and avpops. I'm > >>> using avp_radius to obtain an AVP value from my database > via radius. > >>> What i what to do is replace this value for the RURI. > >>> Here i have a couple of questions. > >>> 1.- The value returned by the avp_radius (the SIP-AVP) where is > >>> stored ? It suppose that the SIP-AVP returned by radius > >> has the form > >>> of "name:value". That "name" refers to the name of what?. > >>> For example i'm returning : > >> "var1:sip:1234567@mydomain.com". What i > >>> see in the debug is : > >>> > >>> avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has > been added > >>> > >>> This is what i got in my ser.cfg (a snippet). > >>> > >>> if (method=="INVITE" || method=="CANCEL") { > >>> if( !avp_load_radius("caller")) { > >>> log (1, "AVP_RADIUS: Fail on > avp_radius\n"); > >>> }; > >> > >>> > >>> if( !avp_pushto("$RURI", "s:var1/g")) { > >>> log (1, "AVPOPS: Fail on AVPOPS\n"); > >>> }; > >>> }; > >>> > >>> Again the debug = 9 . > >>> > >>> 6(23815) avp_load_user: AVP > >> 'var1'='sip:1234567@mydomain.com' has been > >>> added > >>> 6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c: > >>> load_avp_user(344) > >>> 6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from > >>> avp_radius.c: load_avp_user(330) > >>> 6(23815) DEBUG:avpops:pushto_avp: no avp found > >>> 6(23815) AVPOPS: Fail on AVPOPS > >>> > >>> What i'm doing wrong? > >>> Thanks! > >>> > >>> Regards > >>> Ricardo Martinez > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers >

18 years, 11 months

[Serusers] mediaproxy register problem

by Sönmez Güneş

Hi everybody, I setup ser 0.8.14 with mysql support on fedora core 2. And two UA could connect over SER successfully. I give a static IP to my SIP PROXY and followed ONSIP instructions about mediaproxy. But I could not register UAC's behind NAT. begining of my route block: if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("PROXY_IP: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 (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; my register block: # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_contact(); force_rport(); }; if (!www_authorize("PROXY_IP","subscriber")) { www_challenge("PROXY_IP","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; invite block: # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("PROXY_IP","subscriber")) { proxy_challenge("PROXY_IP","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); Any help would be appreciated...

18 years, 11 months

[Serusers] rtproxy based on URI

by Adrian A

Is it possible to configure SER to use RTP proxy only for specific clients, ie ones which have a particular username pattern? I would like to proxy the media only for those phones I have which do not support ICE but support STUN.

18 years, 11 months

[Serusers] dead SIP service providers

by Michael Kendall

Hello All, The following two sites appear to be SIP service providers: www.freeworlddialup.com/ voip.brujula.net/english/messenger-howto.html But when I follow their instruction for setting up Windiows Messenger 4.6, my login attempts tell me, "The service provider is temporarly unavailable, try again later" This makes me suspect that they have shut down their voip servers, and simply failed to delete their web pages. Does anybody know the status of these two companies? They have not responded to my email queries. Freeworlddialup provides a simple tool to evaluate your internet connection before trying their service. The tool consistently claims that my port 5060 is blocked, even though I am directly connected to my cable modem and have all firewalls turned off. It seems like this false result could be caused by an attempt establish a 5060 connection to a non-existant voip server, again supporting my theory that their server has been shut down. Does this theory make sense? Also, both providers say that they work with Windows Messenger 4.6 and 4.7 but not with 5.1. Did Microsoft commit some crime in version 5.1 to make it incompatible with third party SIP service providers? thanks for your help, Michael

18 years, 11 months

[Serusers] missing Digest-Algorithm = "MD5"

by davor jovanovic

Hi all, I'm using SER 0.8.14, FreeRadius 1.0.2. and XLite 1103. My question is about MD5, why I can't see Digest-Algorithm = "MD5" in FreeRadius and XLite client outputs. Is this a problem in XLite or SER? Everything is configured as in HOW TO and I get authenticated with FreeRadius so I suppose that configuration is OK. Dictionaries are the same in radiusclient on SER machine and in FreeRadius. I can provide more outputs if someone is interested to help. Any help would be appreciated. *****This is output from FreeRadius Server****** (no Digest-Algorithm = "MD5" received) ... rad_recv: Access-Request packet from host 161.53.0.131:33854, id=131, length=207 User-Name = "djovanovic.srce" Digest-Attributes = 0x0a11646a6f76616e6f7669632e73726365 Digest-Attributes = 0x0109737263652e6872 Digest-Attributes = 0x022a3432393439336464646235643930376433653433323765333063366566646566336436 6137643933 Digest-Attributes = 0x040d7369703a737263652e6872 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "263ec6802b382ddf8d58b363fd0629ad" Service-Type = Sip-Session Sip-Uri-User = "djovanovic.srce" NAS-IP-Address = 161.53.0.131 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 0 modcall[authorize]: module "preprocess" returns ok for request 0 modcall[authorize]: module "chap" returns noop for request 0 modcall[authorize]: module "mschap" returns noop for request 0 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "djovanovic.srce" Digest-Realm = "srce.hr" Digest-Nonce = "429493dddb5d907d3e4327e30c6efdef3d6a7d93" Digest-URI = "sip:srce.hr" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 0 ... *****This is output from XLite****** (no Digest-Algorithm = "MD5" sent) SEND TIME: 203074718 SEND >> 191.153.206.58:5060 INVITE sip:537@srce.hr SIP/2.0 Via: SIP/2.0/UDP 191.153.206.57:5060;rport;branch=z9hG4bK8BB5DE10CC404EEBABE6761213B4AA45 From: davor <sip:djovanovic.srce@srce.hr>;tag=2361756632 To: <sip:537@srce.hr> Contact: <sip:djovanovic.srce@191.153.206.57:5060> Call-ID: 146A8327-BCC6-42E6-A027-83EA1832631B(a)191.153.206.57 CSeq: 40898 INVITE Authorization: Digest username="djovanovic.srce",realm="srce.hr",nonce="429492e156f0af6f7ae5e821d7 3df9be3e04f360",response="17a54d30cc2d5ecf1cc8e714d40210e0",uri="sip:537@src e.hr" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 308

18 years, 11 months

[Serusers] Call setup delay

by James Peters

I am using SER for VOIP and experiencing issues with call setup when making VOIP to VOIP calls. With 2 end units connected to the same server, there is a 5-7 second delay in the RTP stream when I place a call from one IP phone to another. I also experience a delay in ring back when calling inbound to an IP phone from a standard PSTN line. I am new to configuring SER. I can provide more info if needed. Thanks. James D. Peters Network Engineer NetCarrier, Inc. www.netcarrier.com <http://www.netcarrier.com>

18 years, 11 months

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sr-users May 2005