sr-users November 2003

sr-users@lists.kamailio.org

116 participants
267 discussions

[Serusers] SER/SIP & RADIUS/Auth-Type = Digest
by Madan 12 Nov '03

12 Nov '03

Hi All, I have just completed installation and configuration SER with FreeRadius but I can nt seems to move ahead of the first test phase of the RADIUS-HOWTO doc I have followed all the instructions and can nt really make out as to what is going wrong and where. here is the radius log details, please let me know what is wrong and where modcall[authorize]: module "sql" returns notfound for request 0 modcall: group authorize returns ok for request 0 rad_check_password: Found Auth-Type DIGEST auth: type "digest" modcall: entering group authenticate for request 0 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 0 modcall: group authenticate returns invalid for request 0 auth: Failed to validate the user. Delaying request 0 for 1 seconds Finished request 0 Going to the next request --- Walking the entire request list -- ++users file++ test Auth-Type := Digest, User-Password == "test" Reply-Message = "Hello, test with digest" ++ digest file++ User-Name = "test", Digest-Response = "631d6d73147add2f9e437f59bbc3aeb7",Digest-Realm = "testrealm", Digest-Nonce = "1234abcd " ,Digest-Method = "INVITE", Digest-URI = "sip:5555551212@example.com",Digest-Algorithm = "MD5", Digest-User-Name = "test"

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[Serusers] SER crash with serctl
by alex.zen 12 Nov '03

12 Nov '03

Hi! I have recently downloaded SER-0.8.11 from IPTEL's site and I am having problems using serctl. In fact, I was using a modified version of Ser- 0.8.11pre25 and everything was working fine. I don't know why it stopped working. The problem happens when I try to create an alias using serctl: > serctl alias add 555199 sip:5551100@mydomain Ser just crashes a litlle time after the command get executed. I noted also that the same behaviour happens when adding contacts through serctl, as both additions seems to use the same mechanism (but everything works fine when the contact is added by means of a user agent REGISTER message). The system I'm running SER on is a Connectiva Linux 9 (a Brazilian distribution of Linux based on Red Hat). The kernel version is 2.4.18-2cl. Attached to the message can be found a simplified version of my ser.cfg and the messages generated by SER with debug=9. An output from serctl ps may also help when analyzing the debug messages and is listed below: 0 16399 attendant 1 16401 receiver child=0 sock=0 @ my.IP:5060 2 16402 receiver child=1 sock=0 @ my.IP:5060 3 16403 receiver child=2 sock=0 @ my.IP:5060 4 16406 receiver child=3 sock=0 @ my.IP:5060 5 16407 fifo server 6 16411 timer 7 16412 tcp receiver 8 16413 tcp receiver 9 16417 tcp receiver 10 16418 tcp receiver 11 16419 tcp main process Thanks in advance Alex __________________________________________________________________________ Acabe com aquelas janelinhas que pulam na sua tela. AntiPop-up UOL - É grátis! http://antipopup.uol.com.br/

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Fw: [Serusers] Voicemail
by Raphael Coeffic 12 Nov '03

12 Nov '03

----- Original Message ----- From: "Raphael Coeffic" <coeffic(a)fokus.fhg.de> To: <sesha(a)iic.com> Sent: Wednesday, November 12, 2003 4:30 PM Subject: Re: [Serusers] Voicemail > That means that Sems is not running. > > The voicemail system is composed of Ser doing the SIP job, and Sems > dealing with media. You can get it at developer.berlios.de/projects/sems. > > You must check out a CVS version. Depending on the version of Ser > you are using, you should check out the stable version 'ser_rel_0_8_11' or > the head version if you are using an unstable Ser version. > > -Raphael. > > ----- Original Message ----- > From: "Sesha B" <sesha(a)iic.com> > To: "Klaus Darilion" <darilion(a)ict.tuwien.ac.at>; <serusers(a)lists.iptel.org> > Sent: Wednesday, November 12, 2003 4:10 PM > Subject: RE: [Serusers] Voicemail > > > > I'm getting this error from debugs: > > > > 500 couldn not contact announcement server. > > > > Please let me know what it means? Is it possible to put the voicemail on > the > > PBX? I have the voicemail configured for this number on the PBX. Thank > you. > > > > > > > > -----Original Message----- > > From: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at] > > Sent: Wednesday, November 12, 2003 9:42 AM > > To: sesha(a)iic.com; serusers(a)lists.iptel.org > > Subject: RE: [Serusers] Voicemail > > > > > > Usually, if the phone (user) is registered, the call will be forwarded > > to the SIP phone instead of forwarding it to the voicemail. If a want to > > redirect the call to the voicemail after several seconds (e.g. if no one > > picks up the phone), you have to use the fr_inv_timer timer and set up > > script as shown in: > > http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSING > > > > regards, > > Klaus > > > > > -----Original Message----- > > > From: Sesha B [mailto:sesha@iic.com] > > > Sent: Wednesday, November 12, 2003 2:44 PM > > > To: Jiri Kuthan; serusers(a)lists.iptel.org > > > Subject: RE: [Serusers] Voicemail > > > > > > > > > Hi, > > > > > > I'm trying to implement voicemail. When a call is made to a > > > SIP phone, and > > > if the SIP phone is ringing, the "ngrep port 5060" on the SER > > > shows that it > > > is connected. So, it is looking it as a successful connection > > > and hence it > > > is not going to the answering machine. Is there a roundabout > > > solution for > > > this? Please let me know. Thank you. > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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RE: [Serusers] Voicemail
by Klaus Darilion 12 Nov '03

12 Nov '03

Usually, if the phone (user) is registered, the call will be forwarded to the SIP phone instead of forwarding it to the voicemail. If a want to redirect the call to the voicemail after several seconds (e.g. if no one picks up the phone), you have to use the fr_inv_timer timer and set up script as shown in: http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSING regards, Klaus > -----Original Message----- > From: Sesha B [mailto:sesha@iic.com] > Sent: Wednesday, November 12, 2003 2:44 PM > To: Jiri Kuthan; serusers(a)lists.iptel.org > Subject: RE: [Serusers] Voicemail > > > Hi, > > I'm trying to implement voicemail. When a call is made to a > SIP phone, and > if the SIP phone is ringing, the "ngrep port 5060" on the SER > shows that it > is connected. So, it is looking it as a successful connection > and hence it > is not going to the answering machine. Is there a roundabout > solution for > this? Please let me know. Thank you. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] new version of sip scenario
by Jiri Kuthan 12 Nov '03

12 Nov '03

I'm glad to announce availability of Ray's sip scenario tool version 1.1.11 at http://www.iptel.org/~sipsc/. It is a tool which takes captured SIP messages and generates nice HTML call-flows out of them. Many serusers have told me privately this was a really great tool and I think so too. Thanks again, Ray, -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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RE: [Serusers] Unresolveable destination
by Iizuka, Hirotaka, VF-JP 12 Nov '03

12 Nov '03

Thank you very much for the information, and sorry for the late reply. I want to insert alias="vfrnd.jp" into ser.cfg file, but where exactly should I write in? I was reading the Admin's Guide and found the related part, but it seems I need to type in more than just alias="vfrnd.jp". Here is the copy & pasted document; ______________________________________________________________________ Example 2-5. Use of uri==myself Expression # ser powers a domain "foo.bar" and runs at host sipserver.foo.bar; # Names of served domains need to be stated in the aliases # option; myself would not match them otherwise and would only # match requests with "sipserver.foo.bar" in request-URI alias="foo.bar" alias="sales.foo.bar" route[0] { if (uri==myself) { # the request either has server name or some of the # aliases in its URI log(1,"request for served domain") # some domain-specific logic follows here .... } else { # aha -- the server is not responsible for this # requests; that happens for example with the following URIs # - sip:a@marketing.foo.bar # - sip:a@otherdomain.bar log(1,"request for outbound domain"); # outbound forwarding t_relay(); }; } _______________________________________________________________________________________________________________________ I have a default ser.cfg file. When you stated "add it to your ser.cfg (alias= vfrnd.jp)", referenced from below, did you mean write alias="vfrnd.jp" just above the "if (uri==myself)" line? Hiro -----Original Message----- From: Andrei Pelinescu-Onciul [mailto:pelinescu-onciul@fokus.fraunhofer.de] Sent: Friday, November 07, 2003 7:23 PM To: Iizuka, Hirotaka, VF-JP Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Unresolveable destination On Nov 07, 2003 at 17:21, Iizuka, Hirotaka, VF-JP <hirotaka.iizuka(a)vodafone.com> wrote: > Hello, > > I have a problem with registering to the SER. > I use X-Lite SIP softphone, and when I try to register to the SER, I > get following messages; > [...] > SEND >> 192.168.2.253:5060 > REGISTER sip:vfrnd.jp SIP/2.0 [...] > > > RECEIVE << 192.168.2.253:5060 > SIP/2.0 478 Unresolveable destination (478/TM) [...] vfrnd.jp is not resolvable from the host running ser (neither using DNS SRV or normal DNS A lookups). Try logging in on that host and running host vfrnd.jp or nslookup vfrnd.jp. You should fix your DNS entries or at least add vfrnd.jp to /etc/hosts. Also make sure ser knows that vfrnd.jp is "his" domain (I suppose that's what you want). If when you start ser you don't see vfrnd.jp in the alias list, then add it to your ser.cfg (alias= vfrnd.jp) and restart ser. Andrei

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RE: [Serusers] udp error
by Jiri Kuthan 12 Nov '03

12 Nov '03

see recent Jan's post: -- an example configuration file including nathelper support is in the CVS in sip_router/etc/nathelper.cfg. Note that it requires unstable branch and all the stuff is experimental. Jan. -- At 02:01 AM 11/7/2003, Stephen Miles wrote: >Hi Jiri, > >Thanks for that, makes a lot of sence. I did think thats what we were doing so here is my config file for you to have a look at. > >Could you go through it and tell me where I am going wrong? > >Thanks much, > >Stephen > >route{ > loose_route(); > force_rport(); > fix_nated_contact(); > # 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") { > setflag(1); # remember this is ATA > force_rport(); > fix_nated_contact(); > log("LOG: In NAT clause\n"); > #fix_nated_sdp("3"); > }; > 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); > break; > }; > if (uri=~"portal.commverge.co.nz") { > if (method=="REGISTER") { > # Uncomment this if you want to use digest authentication > if (!www_authorize("portal.commverge.co.nz", "subscriber")) { > www_challenge("portal.commverge.co.nz", "0"); > break; > }; > save("location"); > break; > }; > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > forward(202.180.125.200,5060); > break; > }; > # user on-line...forward to his current destination > forward(uri:host,uri:port); > break; > }; > if (uri=~"202.180.83.14") { > log("LOG: In Portal clause!\n"); > rewritehostport("portal.commverge.co.nz:5060"); > log("LOG: Out of Portal clause!\n"); > }; > > if (!lookup("location")) { > forward(202.180.125.200,5060); > break; > }; > # user on-line...forward to his current destination > forward(uri:host,uri:port); > break; > >} > >route[1] >{ > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; >} > > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Friday, 7 November 2003 12:42 p.m. >To: Stephen Miles >Subject: RE: [Serusers] udp error > > >At 12:15 AM 11/7/2003, Stephen Miles wrote: >>Hi Jiri, >> >>SIP support is not turned on on our pix so it is just acting as a nat device, much the same as a linux box or standard router. > >Than you will need RTP relay. Otherwise the media from public internet >to your natted client will be dropped by the nat box. Also, you will >have to configure ser to send replies symmetricaly (as opposed to >address advertised in Via). Use force_rport for that. > >>What is happening though is that the firewall (pix in this case, cisco 827 in others) is not forwarding/natting the same port as the original request. e.i 192.168.18.12 sends sip message to portal on port 5060, firewall gets that and nats it but uses the next avalible port to send out on so the portal (our sip proxy running ser) gets the request from firewall:next_avalible_port, not firewall:5060. It seems that ser doesn't realise this and thats why we get the error, if I set my softphone (X-Lite) to use the next port the firewall is going to use (from watching logs etc) is works. Sortphone sends register as 192.168.18.12:65389 and the firewall does the same firewall:65389 and it works against ser, phone registers and calls can be make. > >SIP servers are supposed to reply to port number advertised in >rquest's topmost via header field. > >>I hope this makes more sence. If not please let me know as this could be a bit of a show stopper for us using ser and we would very much like to. > >Nevertheless, the rtp relay workaround along with SER configured in a nat-friendly >manner should help you out. > >-jiri -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] RE: [Serdev] SIP "braodcast"
by Jiri Kuthan 11 Nov '03

11 Nov '03

I haven't review the 3gpp stuff in detail but I suppose it takes some SIP extensions which today's SIP phones don't have.r -jiri At 02:16 PM 11/11/2003, Klaus Darilion wrote: >Hi! > >This sounds like a pre-configured conference group, where you dial into >a conference, and then the conference server invites all the other >participants. The problem is, that you have to wait until all >participants have accepted the call before you start speaking. > >Maybe a nicer solution would be the PoC (PTT over Cellular) Service, >which was recently published by several cell phones manufacturer. This >service allows you to use SIP/RTP enabled cell phones in a walky-talky >like mode. Info and the specification can be found at: >http://www.ericsson.com/products/product_selector/Inst_Talk_hpsol.shtml > >regards, >Klaus > > >> -----Original Message----- >> From: Olaf Menzel [mailto:menzel@fokus.fhg.de] >> Sent: Tuesday, November 11, 2003 1:45 PM >> To: Olaf Menzel >> Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org; rco(a)iptel.org >> Subject: [Serdev] SIP "braodcast" >> >> >> >> Hi Jiri, Jan, Raphael and all other Ser users and developers, >> -------------------------------------------------------------- >> >> as I have already mentioned the demining scenario in our >> project there is the need of a Voip SIP "broadcast" >> functionality. This means the resonsable operator should be >> ready to reach the whole demining group >> with one call. Because we do not have real braodcast like VHF >> radio a approach could be the following scenario: >> >> The operator will send a SIP invite to all members of his >> team and when the user ansers the phone call the call will >> automaticly transfered to a SIP conference. This is what I >> have discussed with Raphael. What do you think how to realize >> such a "broadcast" functionality ?? >> >> regards >> >> >> Olaf >> >> >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev >> >> > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] SER && SEMS over satellite
by Olaf Menzel 11 Nov '03

11 Nov '03

Dear SERusers and SERdevelopers, ------------------------------------- For geographic information system we want to use SIP Voip over satellite. The attached figure will give you a short overview of the network architecture. We have a home network and a field network. The field network is directly behind the satellite station. In the field network we will have notebooks and PDA running Voip SIP client software. The VSAT and the WLAN Accesspoint will be mounted in a VAN or other movable vehicle. In the VAN a operation team will control all the proccesses in the field network. For Voip online collaboration they have several Voip hardphones. We'd like to havve Voip bidirectional connection between the operating team and the stuff in the field and Voip conferencing as well. For emergency situations e.g. the field network is a mine field of a demining NGO the operating team should be able to send a emergency call to everybody in the mine filed to stop the demining proccess immediately. Right now demining organizations use VHF radio for this purpose. The operation team in the home office should be able to call everybody in the filed network and participate in the field's Voip conference. To test the Voip functionality I set up a demonstrator network and installed the developement versions of SER and SEMS in the home network and in the field network. Because I am a new in this realm I have additional questions the administrator manual could not answer or I did not unserstand it. 1. Do I need DNS running for SER routing between the home and field network even I am using privat networks ? 2. How and were can I define a conference URI e.g. sip:fieldconf@10.3.240.1 3. The postgresql SER module does not compile under RH 7.3 and RH 9.0 . I cannot find a SQL setup script for postresql. Do I really need a RDMS like MySQL for user management and registration? 4. Does the following scriptlet routes all matching calls from the home network to the field network. ( All numbers beginning with a 2 should be routed to the filed network and all URI'S containing 10.3.240.1 in the domain part: if(uri=~"^sip:2[0-9][0-9][0-9] | (10\.3\.240\.1)" { forward (10.3.240.1, 5060); } All extension beginning with 1 or containing 10.3.1.2 as domain part in the URI should be routed to the SER in the home network. ???? Thank you for your help Olaf -- Fhg-Fokus-CC Satcom

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[Serusers] SIP Hardware Phone
by dhiraj.2.bhuyan＠bt.com 11 Nov '03

11 Nov '03

Greetings All, Can anyone recommend some SIP Hardware Phone that is standards compatible and works with SER? It will be great if you can quote an approximate price for one piece. Thanks in advance, Dhiraj Bhuyan Security Research Engineer BT Exact Tel: +44 1473 643932 Mob: +44 7962 012145 Email: dhiraj.2.bhuyan(a)bt.com

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sr-users November 2003