sr-users April 2005

sr-users@lists.kamailio.org

250 participants
697 discussions

[Serusers] ser modules compilation
by Alex 04 Apr '05

04 Apr '05

I am trying to compile modules under ser. the name of the module is auth_radius. and i am receiving this error. make ../../Makefile.modules:21: "you should run make from the main ser directory" gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"auth_radius.so"' -DVERSION='"0.8.14"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -I/usr/local/include -c authrad_mod.c -o authrad_mod.o authrad_mod.c: In function `mod_init': authrad_mod.c:111: `DICT_VENDOR' undeclared (first use in this function) authrad_mod.c:111: (Each undeclared identifier is reported only once authrad_mod.c:111: for each function it appears in.) authrad_mod.c:111: `vend' undeclared (first use in this function) authrad_mod.c:135: warning: assignment makes pointer from integer without a cast authrad_mod.c:140: too many arguments to function `rc_conf_str' authrad_mod.c:140: too many arguments to function `rc_read_dictionary' authrad_mod.c:145: warning: implicit declaration of function `rc_dict_findvend' authrad_mod.c:163: too many arguments to function `rc_dict_findattr' authrad_mod.c:163: too many arguments to function `rc_dict_findval' make: *** [authrad_mod.o] Error 1 Any help will be appreciated. thanks --------------------------------- Do you Yahoo!? Better first dates. More second dates. Yahoo! Personals

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[Serusers] How to handle second INVITE
by Dirk Willbrandt 04 Apr '05

04 Apr '05

Hi all.. I solved the problem with the forwarding to a PSTN Gateway. I've rewrite the Contact Address of the SIP-Message sented by the Phone with subst(). Now i have a new Problem. It's about accounting with Radius + MySQL. When i start a call, the SIP-Server gets two INVITES because of record_route(). Both of them are written over radius in my MySQL Table but i want only one of them in my MySQL Table and i don't know how to handle that. Have anybody here an Idea to this problem ? Dirk

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Re: [Serusers] Nathelper/Rtpproxy not working for two natted clients
by Greger V. Teigre 04 Apr '05

04 Apr '05

Dear Vivienne, The idea behind the ONsip.org Getting Started document is to create one source for a thorough coverage of the aspects that tend to arise for most users. It's an introduction to the conceptual underpinnings of SER, as well as a reference design and corresponding configuration files that can be used to quickly implement the functionality that most users will want. I wrote section 1 in that document, and I suggest that you read it. If you have questions to what is written there or find that certain things are not covered well enough, I will be happy to answer your questions. As an added bonus, you will make me see where the text is confusing or where more coverage is needed and the document will be improved. I understand that it is easier to ask a quick question on serusers and get an answer, but we hope that the Getting Started document will bring everybody quickly up to speed in the basic stuff, so that we can concentrate on exploiting more advanced aspects of SER and stimulate to further development. i hope you will find reading the Getting Started document rewarding! Thanks, Greger ---- Original Message ---- From: Vivienne Curran To: Greger V. Teigre Sent: Monday, April 04, 2005 03:06 PM Subject: Re: [Serusers] Nathelper/Rtpproxy not working for two natted clients > Hi Greger, > > Can you can give me an idea of how to handle the callers? If I can > modify my script for this functionality do you think I am nearly > there? My script must handle natted clients on separate networks and > also natted clients on the same subnet. I want all the calls proxied > for the moment. > > Thanks again, > Vivienne > > "Greger V. Teigre" <greger(a)teigre.com> wrote: > Vivienne, > From your config: > if (nat_uac_test("3")){ > if (method == "REGISTER" || ! > search("^Record-Route:")){ > log("Log: Someone trying to register from > private IP,rew > riting\n"); > fix_nated_contact(); #Rewrite contact with source IP > 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 > }; > > Here you don't proxy the call, you just add the direction=active. If > you want to rtp proxy the calls, you need to have a force_rtp_proxy() > for the initial INVITE. Read section 1 of the Getting Started > document at ONsip.org to understand what is supposed to happen and > how to read the SDP payload and see the rtpproxy example found in the > document. > Your config will process the OK and proxy the rtp stream from the > callee, but not the caller's. > g-) > > ---- Original Message ---- > From: Vivienne Curran > To: serusers(a)lists.iptel.org > Sent: Thursday, March 31, 2005 05:55 PM > Subject: [Serusers] Nathelper/Rtpproxy not working for two natted > clients > >> Hi, >> >> I am having problems troubleshooting a problem I am experiencing with >> my SER configuration. I have ser 0.8.14 running with rtpproxy and >> nathelper enabled. I have two phones on the same subnet behind nat >> and I would like to make a call between the two. I want to invoke >> rtpproxy for this as they both have private address [I know this >> isn't the most efficient way as they're both on the same subnet but I >> can worry about that later]. >> >> When I ring from the phone 1 (2092) to phone 2 (2093), 2092 can hear >> voice but 2093 can't. When 2093 ring 2092, there's no audio. These >> phones are Grandstream BT100's. They have been configured to listen> >> on different SIP and RTP ports. >> >> 2092: SIP Port: 5060 >> 2092: RTP Port: 5004 >> 2093: SIP Port: 5061 >> 2093: RTP Port: 5005 >> >> I have included my ser.cfg file the messages received on my SER >> server (using ngrep SIP-q)in an attachment. I can confirm that my >> rtpproxy is working (originally I thought it wasn't) by using "strace >> -d <rtpproxy pid> -f -F". I can see a signal being returned. >> >> Any help would be appreciated or advise as to how I can proceed >> troubleshooting. >> Kindest Regards, >> Vivienne. >> Send instant messages to your online friends >> http://uk.messenger.yahoo.com >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > Send instant messages to your on! line friends > http://uk.messenger.yahoo.com

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[Serusers] Child procs
by Matt Schulte 04 Apr '05

04 Apr '05

Do increasing child procs actually do anything? When I do a 'ps axu' I only show 4 of the procs having CPU time, the rest having no cpu time at all. Any thoughts on this? Thanks, Matt

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Re: [Serusers] asterisk or ser
by harry gaillac 04 Apr '05

04 Apr '05

Hello, Look at http://www.sipfoundry.org/ for sipxpbx it's a feature-reach ipbx Harry --- "T. Maron" <t_maron_lists(a)yahoo.com> wrote: > Hello, > > I was playing with ser and asterisk for a few weeks > but I couldn't make a decision what is better to use > for medium voip network (more than 500 terminals in > several locations, sepatate pstn gw, basic services > - > voice and voicebox). Asterisk looks like a very > feature-rich voip application and ser more flexible. > > Can anyone share some experiences deploying either > of > the two voip systems? The maintenance effort is what > weight more for us, we are looking for qualified > support to install the platform once we are decided > what to use. > > > Many thanks in advance, > TM > > PS. one can write me directly if the content of the > mail is intended to be private. > > > > __________________________________ > Do you Yahoo!? > Make Yahoo! your home page > http://www.yahoo.com/r/hs > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/

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[Serusers] Nathelper/Rtpproxy not working for two natted clients
by Vivienne Curran 04 Apr '05

04 Apr '05

Hi, I am having problems troubleshooting a problem I am experiencing with my SER configuration. I have ser 0.8.14 running with rtpproxy and nathelper enabled. I have two phones on the same subnet behind nat and I would like to make a call between the two. I want to invoke rtpproxy for this as they both have private address [I know this isnt the most efficient way as theyre both on the same subnet but I can worry about that later]. When I ring from the phone 1 (2092) to phone 2 (2093), 2092 can hear voice but 2093 cant. When 2093 ring 2092, theres no audio. These phones are Grandstream BT100s. They have been configured to listen on different SIP and RTP ports. 2092: SIP Port: 5060 2092: RTP Port: 5004 2093: SIP Port: 5061 2093: RTP Port: 5005 I have included my ser.cfg file the messages received on my SER server (using ngrep SIP-q)in an attachment. I can confirm that my rtpproxy is working (originally I thought it wasnt) by using strace d <rtpproxy pid> -f F. I can see a signal being returned. Any help would be appreciated or advise as to how I can proceed troubleshooting. Kindest Regards, Vivienne. Send instant messages to your online friends http://uk.messenger.yahoo.com

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[Serusers] binding expired
by Wojciech Ziniewicz 04 Apr '05

04 Apr '05

[...] Binding '497','sip:497@192.168.0.136:5060' has expired [...] Any ideas how to stop sipphones making theese logs ? -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535

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[Serusers] ul_add: flags expected
by Steve Blair 04 Apr '05

04 Apr '05

Hello: I am getting the 'ul_add: flags expected' error with v0.9.0 and serctl. There is some chatter in the archives about the perm column in the subscriber table but it is unclear to me if this is how to fix the problem? Any ideas? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] RFC3261 transaction matching failed
by Mimmus 04 Apr '05

04 Apr '05

Forwarding a call from Asterisk to Microsoft Live Communication Server 2005 via SER (to translate from UDP to TCP), I get a 'one-way' communication (WMessenger user can hear voice but PSTN phone user cannot). Running SER in debug mode, I found: DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found Searching Google, I found that this is a known bug in Asterisk. I upgraded to 1.0.7 (Debian unstable) but bug is still present. Peraphs, Asterisk mailing-list would be a better choice for this message but anyway can someone confirm this? Thanks Domenico Viggiani

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[Serusers] What is the MySQL uri Table For?
by Java Rockx 04 Apr '05

04 Apr '05

Hi All. What is the MySQL uri table use for? Regards, Paul

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sr-users April 2005