sr-users March 2005

sr-users@lists.kamailio.org

250 participants
627 discussions

Re: [Serusers] Kerberos + SER?
by Samuel Osorio Calvo 25 Mar '05

25 Mar '05

I think I saw somebody posting some patches to use TLS under GPL with SER....just dig into the archives and you'll get a surprise. Samuel. Unclassified. >>> "Justin Pearce" <Justin(a)PriceVideo.com> 03/24/05 11:36PM >>> Hello all, I am wondering if it is possible to use Kerberos with SER to get secure user Authentication? I know that TLS would be best, however SER does not yet impliment TLS in freely available releases. Any thoughts on the matter? Anyone have this working in the field? Thanks, Justin

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[Serusers] Kerberos + SER?
by Justin Pearce 24 Mar '05

24 Mar '05

Hello all, I am wondering if it is possible to use Kerberos with SER to get secure user Authentication? I know that TLS would be best, however SER does not yet impliment TLS in freely available releases. Any thoughts on the matter? Anyone have this working in the field? Thanks, Justin

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[Serusers] CDRTool with SER
by Jed Stafford 24 Mar '05

24 Mar '05

I have been working with CDRTool now for a couple of days trying to get it setup. I've finnaly got SER writing to Freeradius, etc. Now, when I try to get in to CDRTool I get past the login screen and am stuck with this error. Fatal error: Cannot instantiate non-existent class: cdrs_ser_radius in /CDRTool/callsearch.phtml on line 58 In my global.inc file I do have this class to my knowledge. I trimmed out all of the options in the array to the bare bones as below and still get the error. Any ideas? The documentation is fairly light for the tool. You would think it would be the opposite since this is supposed to convince you to purchase a license and get hooked. $DATASOURCES=array( "ser_radius" =>array( "name" => "SIP Express Router (Radius Accounting) ", "class" => "CDRS_ser_radius" ), Thanks! -J

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Re: [Serusers] SER x Nextone problem
by Java Rockx 24 Mar '05

24 Mar '05

Maybe ACKs aren't being record-routed. Regards, Paul On Thu, 24 Mar 2005 10:36:02 -0300, Rodrigo P. Telles <telles(a)devel-it.com.br> wrote: > Hi, > > Java Rockx escreveu: > > It kind of sounds like you're not sending an ACK for a 200OK response > > from the Nextone box. > > Hummm, but SER should send this ACK from 200OK for everyone or is it only for > Nextone? > I never had this problem with other equipments. > > > > > Can you send an ngrep dump of the sip call? > > Yes I can, I will do that. > Thanks for your answer. > > > > > > Regards, > > Paul > > > > > > On Fri, 18 Mar 2005 13:22:04 -0300, Rodrigo P. Telles > > <telles(a)devel-it.com.br> wrote: > > > >>Hi guys, > >> > >>I've been experiencing some troubles with carrier's that are using > >>Nextone (www.nextone.com). > >>I could place calls through Nextone but no longer then 1 minute, between 48 and > >>53 seconds the call is dropped with a BYE sent by Nextone. > >>I've been using another carriers using Cisco and Quintum without problems. > >>Does some one experimenting this problem? > >> > >>I'm using SER 0.8.14. > >> > >>Thanks in advance. > >> > >>-- > >>============================================ > >>Rodrigo P. Telles <telles(a)devel.it> > >>IVOZ # 1009 > >>Project Manager > >>Devel-IT - http://www.devel.it > >>Bestcom Group > >>============================================ > >> > >>_______________________________________________ > >>Serusers mailing list > >>serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > > -- > ============================================ > Rodrigo P. Telles <telles(a)devel.it> > IVOZ # 1009 > Gerente de Projetos > Devel-IT - http://www.devel.it > Grupo Bestcom > ============================================ >

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[Serusers] SEMS conferencing - use GSM codec
by Steven Langley 24 Mar '05

24 Mar '05

Hi there SEMS seems to use one of the g.711 codecs by default, which uses quite a bit of bandwidth. I believe that SEMS also ships with GSM, which uses a lot less bandwidth. I am wondering where I would specify that I want to use this codec instead of the default one? Many thanks Steven

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[Serusers] NAT and forward to PSTN
by John Dante 24 Mar '05

24 Mar '05

Hi! SER is running in my Fedora Core 3 box. I've registered in my SER two IP phones located behind NAT, used the example nathelper.cfg script. How could I manage to forward calls to a PSTN gateway, example all calls beginning by "0..." ? What are the script modifications? Thanks john

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[Serusers] SEMS conference monitoring
by Steven Langley 24 Mar '05

24 Mar '05

Hi I am wondering if there is some utility (either with SEMS or third party) that can be used to monitor conferences and the users in those conferences? Otherwise is there an alternative if want to do this? Many thanks Steven

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[Serusers] BYE, nat, Xlite
by Giovanni Balasso 24 Mar '05

24 Mar '05

Hi serusers! After setting up an almost working ser with nat, rtpproxy and asterisk ivr, I'm stuck on an hangup problem call if from behind nat. I can't get xlite BYEs to hit ser (ngrep on ser machine gets nothing on hangup pressed on client). The address in SEND>> is ser nat internal address and I'm calling from another nat. This is xlite log on BYE SEND TIME: 735130 SEND >> 192.168.1.100:5060 BYE sip:asterisk@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bKE406E7BA9C6E11D98395000272430644 From: gio <sip:gio@SER_PUBLIC_IP>;tag=1837372280 To: <sip:gio@SER_PUBLIC_IP>;tag=as0d4fbf68 Contact: <sip:gio@192.168.1.5:5060> Route: <sip:192.168.1.100;ftag=1837372280;lr=on> Call-ID: D6C66B0C-9C6E-11D9-8395-000272430644(a)192.168.1.5 CSeq: 8745 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 where 192.168.1.100 is private ip address of ser and asterisk server and 192.168.1.5 is xlite private address from another private network. I'm pretty sure the problem is SEND >> 192.168.1.100 shoul be SEND >> SER_PUBLIC_IP. Reading past issues I tried fiddling with fix_rport, fix_nated_contact, but no way. Can anyone please explain the correct way to route BYEs thru NAT? Other things work: i can hear sounds, talk to other people and record messages in asterisk boxes. Please, advice Thanx everyone In my ser.cfg commands involved in BYE and nat are the following: if (nat_uac_test("1")) { if (method == "REGISTER" || !search("^Record-Route:")) { fix_nated_contact(); if (method == "INVITE" || method == "BYE" || method == "CANCEL") { # I think I could remove method == BYE and CANCEL with no harm... fix_nated_sdp("1"); }; force_rport(); setflag(6); }; }; record_route(); if (loose_route()) { route(1); }; if (method=="INVITE") { if (lookup("location")) { setflag(1); route(1); break; }; }; route[1] { if (isflagset(6)){ force_rtp_proxy(); t_on_reply("1"); }; if (method == "BYE" || method == "CANCEL"){ setflag(1); }; if (!t_relay()){ sl_reply_error(); }; break; } ciao -- Giovanni Balasso giaso(a)yahoo.it

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[Serusers] Do Call but Can't Receive Calls (NAT & Port Forwarding)
by Ozan Blotter 24 Mar '05

24 Mar '05

I'm using SER on a Public IP, do IPv4 and IPv6 Conversion, Nathelper Module and RTPProxy. Both parties are behind NAT and gateways/routers arranged to forward 5060 UDP and TCP ports to LAN IP's of the ATA's on both side. There are 3 Registered Users, 8334454556 can call 8334843600, everything is okay, each party talks, the quality of voice is great. But when 8334843600 calls 8334454556, there's no ringing tone, nothing happens. When i check wih serctl util, i realized one thing like ; Contact : 'sip:8334454556@81.215.239.231:5060' Call-ID : '13491B1111341212(a)81.215.239.231' <<< This is Public IP Contact : 'sip:8334843600@85.96.192.140:5060' Call-ID : '1360D1B911761212(a)192.168.0.101' <<< This is LAN IP What may be wrong with SER or ATA's. root@sipproxy~> serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'location_inet6' size : 512 table: 0x422b80f8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location_inet4' size : 512 table: 0x422b6090 d_ll { n : 3 first: 0x422ba1f8 last : 0x422ba100 } ...Record(0x422ba1f8)... domain: 'location_inet4' aor : '103201' ~~~Contact(0x422ba328)~~~ domain : 'location_inet4' aor : '103201' Contact : 'sip:103201@212.156.169.125:5060' Expires : 1491 q : 0.00 Call-ID : '7320E5ADBA8E1526(a)212.156.169.125' CSeq : 70 replic : 0 User-Agent: 'Unknown' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0x422ba2e0)... domain: 'location_inet4' aor : '8334454556' ~~~Contact(0x422ba238)~~~ domain : 'location_inet4' aor : '8334454556' Contact : 'sip:8334454556@81.215.239.231:5060' Expires : 56 q : 0.00 Call-ID : '13491B1111341212(a)81.215.239.231' CSeq : 128 replic : 0 User-Agent: 'Unknown' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0x422ba100)... domain: 'location_inet4' aor : '8334843600' ~~~Contact(0x422ba148)~~~ domain : 'location_inet4' aor : '8334843600' Contact : 'sip:8334843600@85.96.192.140:5060' Expires : 32 q : 0.00 Call-ID : '1360D1B911761212(a)192.168.0.101' CSeq : 8 replic : 0 User-Agent: 'Unknown' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== root@sipproxy~> -- >>> please do not use >>> l.s.d(a)dnaofdesign.com >>> above account is stand by >>> use same account as >>> cosmocid(a)ispro.net.tr

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[Serusers] Restrict registration to one user per login
by Jffdsaklj dsjk 24 Mar '05

24 Mar '05

How can we restrict the registration to only one user per login?. Currently, you can register with the same login/password in different machines at the same time. But what we require is that only the first registration is valid and the rest ones are invalidated by the SER. Thanks

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sr-users March 2005