sr-users March 2004

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145 participants
286 discussions

[Serusers] Help needed : SER / PSTN / NAT
by olivier＠siteboulevard.com 18 Mar '04

18 Mar '04

Hi, After some testing on the latest release, i have some problem doing the following on LINUX : Scenario : - SIP Phones behind a NAT - SER server under linux with rtpproxy launched - a 3660 cisco gateway with PSTN connectivity enabled. When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere. I could not find a sample ser.cfg script that reflect this scenario. Could someone send me this scenario ? Maybe i missunderstood some things. In particular, do i need to launch two instances of ser (one for outbound proxy, another for request. If yes, how to do that) Thanks.

2 6

RE: [Serusers] authentication recommnedations
by Ticknor.Scott＠ic.gc.ca 18 Mar '04

18 Mar '04

> -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: Thursday, March 18, 2004 10:38 am > To: Ticknor, Scott: DGSE > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] authentication recommnedations > > > Yes, there is an explanation. The default configuration file > authenticates REGISTERs only. If you want to authenticate calls > (INVITEs) as well, then you have to modify the configuration file and > insert additional authentication when INVITE is being processed. > > But be careful -- you cannot authenticate all INVITE messages because > foreing users will be not able to call your subscribers (foreign users > have no username and password on your server). You should check if the > INVITE is coming from one of your subscribers and then authenticate. > > Jan. ah ok so i have to play with the routing logic, good good.this will be fun merci tout le monde

1 0

RE: [Serusers] authentication recommnedations
by Ticknor.Scott＠ic.gc.ca 18 Mar '04

18 Mar '04

> -----Original Message----- > From: Nils Ohlmeier [mailto:nils@iptel.org] > Sent: Thursday, March 18, 2004 10:38 am > To: serusers(a)lists.iptel.org > Cc: Ticknor, Scott: DGSE > Subject: Re: [Serusers] authentication recommnedations > > > You have to activate authentication for all type message by > hand, like you > allready did for REGISTER by uncommenting the lines in the config. > Without seeing your config and watching into my crystal ball > i guess you > configuration only authenticates REGSITER messages. RADIUS > will help you in > no way, because you also have to place the autentication > parts at the right > places of your config. > > Nils ok i think i see what you mean , ie: if (method=="INVITE") { www_challenge("sip.foo.bar", "0"); .... } ? i will give this a shot thanks nils

1 0

[Serusers] serweb
by gaillac harry 18 Mar '04

18 Mar '04

Hello I have successfully installed and configured Serweb but cannot login to the web site. I enter admin / heslo and the web page just refreshes. I am not logged in. Can some one point me in the right direction? I am completely stumped. "register_globals=On" and "short_open_tag=On" in /etc/php,ini My configuration: Redhat Taroon updated ser-0.8.12-0 sems-0.1.1-0 php-4.3.2-7.ent mysql-3.23.56-1.9 mysql-server3.23.56-1.9 php-mysql-4.3.2-7 httpd-2.0.46-26.ent serweb_2004-01-04.tar.gz Best regards Harry

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[Serusers] UA behind symmetric NAT advertising Public address
by zeusng 18 Mar '04

18 Mar '04

Hi, Is that a way to check the Top most Via rport with the sender's port? I have a UA capable of detecting its symmetric NAT's public address and advertise it in the SIP message. Below is an example of the message. The device has private address 192.168.x.x and is set to listen SIP message at port 5070, the NAT's public address is y.y.y.y, the register is listening at public address z.z.z.z port 5060. UA1 192.168.5.5:5070 --> NAT (y.y.y.y:60500) --> Proxy z.z.z.z:5060 SEND >> z.z.z.z:5060 REGISTER sip:xyz.org SIP/2.0 Via: SIP/2.0/UDP y.y.y.y:5070;rport;branch=z9hG4bKA38B122163B34FA9B039C915AA2ACB43 From: UA1 <sip:ua1@xyz.org> To: UA1 <sip:ua1@xyz.org> Contact: "UA1" <sip:ua1@y.y.y.y:5070> Call-ID: F75DAF8081B44B62905C14B20CC7421C(a)xyz.org CSeq: 7900 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: Some SIP User Agent Content-Length: 0 RECEIVE << z.z.z.z:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP y.y.y.y:5070;rport=60500;branch=z9hG4bK381E2433E71D4FE995B208FFD8AF8F15 From: UA1 <sip:ua1@xyz.org> To: UA1 <sip:ua1@xyz.org>;tag=794fe65c16edfdf45da4fc39a5d2867c.8eac Contact: "UA1" <sip:ua1@y.y.y.y:5070> Call-ID: F75DAF8081B44B62905C14B20CC7421C(a)xyz.org CSeq: 7900 REGISTER Contact: <sip:ua1@y.y.y.y:5070>;q=0.00;expires=1800 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 z.z.z.z:5060 "Noisy feedback tells: pid=18470 req_src_ip=y.y.y.y req_src_port=60500 in_uri=sip:xyz.org out_uri=sip:xyz.org via_cnt==1" Here, the client 192.168.5.5 (not show in the message) send a message with source address 192.168.5.5 source port 5070 to proxy z.z.z.z port 5060. Because of some intellegence in the UA (STUN in this case), it determines that the public address of the NAT is y.y.y.y. When contructing the Register request, it use the public IP y.y.y.y in the Via and Contact header. The message, upon reaching the NAT, be translated to have source address y.y.y.y and port 60500. None of the nat_uac_test() function can detect that the UA is behide NAT and thus save the contact as sip:y.y.y.y:5070. The reply of the register is fine as it follow the rport. If however, someone invite this client, SER will return sip:y.y.y.y:5070. Due to the nature of symmetric NAT, port 5070 is block and cannot reach UA1. I know I can setup PAT to have all incoming traffic with port 5070 to forward to 192.168.5.5 and make is looks like restricted cone NAT but the NAT device is own by the customer and I have no control. Is there a way in SER that I can check the rport with the source port, if different, rewrite the contact as sip:[source address]:[rport]? Regards, Zeus Ng, CISSP, CCSA Principal Consultant iSquare Technology Tel: +61 2 9419 3887 Fax: +61 2 9410 2629 Mob: 0416 135 794 Email: zeus.ng(a)isquare.com.au ********************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error, you are prohibited from reading, copying, distributing and using the information. Please contact the sender immediately by return email and destroy the original message. ******************************************************************

4 4

Re: [Serusers] SER and rtpproxy
by Klaus Darilion 18 Mar '04

18 Mar '04

I can't help you with the cvs version as I never used it (and it will cause problems as its under heavy developement). If you don't need IPv4-IPv6 translation and separation of rtpproxy from ser, than I suggest to use 0.8.12 stable and the corresponding rtpproxy. ethereal: just sniff all packets without filter. You should see the RTP streams sent to the IP address and port numbers in the SDP of INVITE and 200 OK. Watch for ICMP errors. Klaus Fábio Silvestri wrote: > I'm using cvs version of ser on outbound proxy, and tgz of rtpproxy cvs > version of rtpproxy, I got 'rtpproxy syntax error' messages. > > Klaus, what kind of information on ethereal I need to look? > > > > Klaus Darilion wrote: > >> You have to use rtpproxy v20040105 if you're working with 0.8.12 stable. >> >> What do you mean with "both clients behind a proxy"? The clients are >> registered at different proxies? Then only one proxy has to enable the >> rtpproxy. >> >> klaus >> >> Fábio Silvestri wrote: >> >>> After cvs update on rtpproxy, now I'm getting this message when running: >>> >>> # rtpproxy -2f: >>> >>> rtpproxy: rtpproxy started, pid 1681 >>> rtpproxy: command syntax error >>> rtpproxy: command syntax error >>> rtpproxy: command syntax error >>> rtpproxy: command syntax error >>> rtpproxy: command syntax error >>> >>> >>> >>> Fábio Silvestri wrote: >>> >>>> Hi! >>>> >>>> I have a outbound proxy configured and working very well, but I have >>>> a little problem, when both clientes are registered on SER and >>>> behind a proxy! >>>> >>>> The call's don't work, or the phone rings one time only, and in all >>>> cases the return voice, can't be listen. >>>> >>>> Klaus, i'm using a script I have founded from you! >>>> >>>> Regards. >>>> >>>> Klaus Darilion wrote: >>>> >>>>> Hi Fabio! >>>>> >>>>> Just replace your condition in the reply route with my condition. >>>>> >>>>> No, you don't need to any special commands for the rtpproxy. >>>>> >>>>> Yes I have sample cfg - i already postet is ome time ago - search >>>>> the archive. >>>>> >>>>> regards, >>>>> klaus >>>>> >>>>> PS: cc to the list >>>>> >>>>> Fabio Silvestri wrote: >>>>> >>>>>> Well, where do I need to include that code if { } on my ser.cfg? >>>>>> >>>>>> Do you have a sample cfg? >>>>>> >>>>>> How do I need to run rtpproxy, have some special command line? >>>>>> >>>>>> On Mon, 1 Mar 2004, Klaus Darilion wrote: >>>>>> >>>>>> >>>>>>> Date: Mon, 01 Mar 2004 09:35:12 +0100 >>>>>>> From: Klaus Darilion <klaus.mailinglists(a)pernau.at> >>>>>>> To: Fábio Silvestri <fabio(a)informatec.com.br> >>>>>>> Cc: serusers(a)lists.iptel.org >>>>>>> Subject: Re: [Serusers] SER and rtpproxy >>>>>>> >>>>>>> >>>>>>> >>>>>>> Fábio Silvestri wrote: >>>>>>> >>>>>>> >>>>>>>> Hi! >>>>>>>> >>>>>>>> After a long search on maillinst, I'm trying to run SER with >>>>>>>> rtpproxy, for resolve problems with SIP UA (ata186) behind a >>>>>>>> nat, but I'm still get in trouble. >>>>>>>> >>>>>>>> I have setup ser.cfg to use rtpproxy, like this: >>>>>>>> >>>>>>>> ... >>>>>>>> if (method=="INVITE") { >>>>>>>> log(1,"INVITE\n"); >>>>>>>> fix_nated_sdp("1"); >>>>>>>> record_route(); >>>>>>>> force_rport(); >>>>>>>> setflag(1); >>>>>>>> force_rtp_proxy(); >>>>>>>> t_on_reply("1"); >>>>>>>> } >>>>>>>> ... >>>>>>>> onreply_route[1] { >>>>>>>> log(1, ">=====> reply 1\n"); >>>>>>>> if ((status=~"[12][0-9][0-9]")) { >>>>>>>> force_rtp_proxy(); >>>>>>>> } >>>>>>>> } >>>>>>>> ... >>>>>>>> >>>>>>>> But I get this error on syslog: >>>>>>>> >>>>>>>> 0(5463) ERROR: extract_mediaip: no `c=' in SDP >>>>>>>> >>>>>>> >>>>>>> This will happen when you try to rewrite a response without sdp, >>>>>>> e.g. 100 trying, 180 ringing.... >>>>>>> >>>>>>> I'm unsing: >>>>>>> if ( (status=~"200" || status=~"183") ) { >>>>>>> force_rtp_proxy(); >>>>>>> >>>>>>> >>>>>>>> I'm running rtpproxy, on linux shell without any parameters. >>>>>>>> >>>>>>>> Does anybody knows what this means that message, and how to >>>>>>>> solve, a ser.cfg for rtpproxy, and others tips will be helpfull... >>>>>>>> >>>>>>>> Regards. >>>>>>>> >>>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>> >>> >> >

2 1

[Serusers] Future of ser - still open source?
by Klaus Darilion 18 Mar '04

18 Mar '04

Hi! There is an article about VoIP at heise.de (http://www.heise.de/newsticker/meldung/45655) which announces that the next releases of ser won't be open source. Can please comment on this? regards, Klaus

4 5

[Serusers] ldat and ser
by Helmut Herzog 18 Mar '04

18 Mar '04

Hello, I have heard, that You have a ldap authentification for SER. I want to authenticate against a Windows ADS. Can You say me, how to get this? -- Helmut Herzog Fraunhofer Institut für Informations- und Datenverarbeitung IITB Fraunhoferstr. 1 D-76131 Karlsruhe Tel: 0049-721/6091216 Fax:0049-721/6091413 http://www.iitb.fraunhofer.de/ email: helmut.herzog(a)iitb.fraunhofer.de

1 0

[Serusers] SER and Asterisk Voicemail with MWI?
by Tor Setane 18 Mar '04

18 Mar '04

This works for me with Cisco 7960 phones, asterisk sip.conf for #8544: [8544] type=friend username=8544 secret=**** host=dynamic defaultip=10.100.10.118 (ip of ser) mailbox=8544 Paul Cheng wrote: >Hi, > >Has anyone successfully used Asterisk as a back end voicemail server >with SER and figured out a way to pass MWI messages to the client UA >registered with SER? regards, Tor.

1 0

[Serusers] SER and Asterisk Voicemail with MWI?
by Paul Cheng 17 Mar '04

17 Mar '04

Hi, Has anyone successfully used Asterisk as a back end voicemail server with SER and figured out a way to pass MWI messages to the client UA registered with SER?

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sr-users March 2004