Dual homed server, operating as a vpn server, has one public and one private ip.
Remote client pc's ( scattered around the internet ) connect to server's private ip
(via openvpn) tunnel, server NATs traffic originating from private/vpn interface
through server's public ip off to the internet at large; all this works fine: pc's route
all traffic through vpn.
Now, if this server is also running an instance of ser, via which the client pc's
register - therefore, the NAT'ing takes place on the server _itself_, rather than
on some intermediary server/router _between_ the pc's and ser server - is it
still necessary to run mediaproxy or nathelper?
I'm still cutting my teeth on voip in general, and ser in particular - however our
environment seems quite different to what I have seen and read in all the
docs and material regarding sip/ser and the NAT issues involved -- all of these
seem to assume that the NAT'ing takes place on external/3rd-party servers/routers
rather than on the ser server itself.
As such, I'm having all kinds of fun trying to figure out what the heck I'm doing -
whether I'm on the right track, and am merely suffering from some small misconfig
somewhere that is preventing all this to work, or whether this scenerio is so outlandish
that I'm unlikely to ever succeed period.
So what I'm hoping to get, is some indication from you experts on whether this
environment I've tried to conceptualize above is possible or not; and any tips or
suggestions you may have. I'll be happy to provide further clarification on anything
that I haven't described clearly.
Many thanks!
Cheers,
Corey
Having some troubles with a new voip gateway service we're using; hoping
someone could provide any advice.
To sum up:
Call is initiated from pots line; gets routed to voip gateway provider; gateway
forwards invite to our ser server; one ring then busy signal is heard on
originating/requesting pots line; the destined softphone UA, registered via
our ser server, never gets call.
As far as I can fathom, ser recieves an INVITE forwarded/routed from the
gateway, then ser requests auth/cred, then gateway responds with an
ACK and at that point everything just stops.
Assistance is much appreciated, as I'm unable to determine what may be
causing the issue.
208.xxx.xxx.xx = 3rd-party/commercial voip gateway server
206.xx.xx.x = our ser proxy server
1234567890 = originating pots line
8889997777 = softphone UA number ( aliased to sip uri 'blah(a)206.xx.xx.x' )
ngrep:
#
U 208.xxx.xxx.xx:5060 -> 206.xx.xx.x:5060
INVITE sip:8889997777@206.xx.xx.x SIP/2.0..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To: <s
ip:8889997777@206.xx.xx.x>..Contact: <sip:1234567890@208.xxx.xxx.xx>..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70
..Date: Fri, 24 Feb 2006 22:28:05 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Content-Type: application/sdp..Content-Length: 397....v=0..o=root 32669 32669 IN IP4
208.xxx.xxx.xx..s=session..c=IN IP4 208.xxx.xxx.xx..t=0 0..m=audio 10784 RTP/AVP 0 18 8 3 111 4 97 101..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:8 PCMA/80
00..a=rtpmap:3 GSM/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
#
U 206.xx.xx.x:5060 -> 208.xxx.xxx.xx:5060
SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport=5060..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To
: <sip:8889997777@206.xx.xx.x>;tag=2d38da4c38ea547b795405b32554c3bb.f3cb..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 INVITE..Proxy-Authenticate: Digest realm="208.1
39.204.245", nonce="43fe339b33d7c58a0d89e4711838584d"..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 206.xx.xx.x:5060 "Noisy feedback tells: pid=2
974 req_src_ip=208.xxx.xxx.xx req_src_port=5060 in_uri=sip:8889997777@206.xx.xx.x out_uri=sip:8889997777@206.xx.xx.x via_cnt==1"....
#
U 208.xxx.xxx.xx:5060 -> 206.xx.xx.x:5060
ACK sip:8889997777@206.xx.xx.x SIP/2.0..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To: <sip:
8889997777(a)206.xx.xx.x>;tag=2d38da4c38ea54795405554c3bb...Contact: <sip:1234567890@208.xxx.xxx.xx>..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 ACK..User-A
gent: Asterisk PBX..Max-Forwards: 70..Content-Length: 0....
... end of transmission;
no further exchange after gateway ACK;
originating pots line hears a single ring, then fast busy signal;
softphone UA never recieves INVITE or anything
ser debug:
IP Request:
0(2974) method: <INVITE>
0(2974) uri: <sip:8889997777@206.xx.xx.x>
0(2974) version: <SIP/2.0>
0(2974) parse_headers: flags=1
0(2974) Found param type 232, <branch> = <z9hG4bK40f0830c>; state=6
0(2974) Found param type 235, <rport> = <n/a>; state=17
0(2974) end of header reached, state=5
0(2974) parse_headers: Via found, flags=1
0(2974) parse_headers: this is the first via
0(2974) After parse_msg...
0(2974) preparing to run routing scripts...
0(2974) parse_headers: flags=128
0(2974) end of header reached, state=9
0(2974) DEBUG: get_hdr_field: <To> [31]; uri=[sip:8889997777@206.xx.xx.x]
0(2974) DEBUG: to body [<sip:8889997777@206.xx.xx.x>
]
0(2974) get_hdr_field: cseq <CSeq>: <102> <INVITE>
0(2974) DEBUG:maxfwd:is_maxfwd_present: value = 70
0(2974) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16
0(2974) DEBUG: add_param: tag=as5f35b551
0(2974) end of header reached, state=29
0(2974) parse_headers: flags=256
0(2974) DEBUG: get_hdr_body : content_length=397
0(2974) found end of header
0(2974) find_first_route: No Route headers found
0(2974) loose_route: There is no Route HF
0(2974) is_local(): Realm '206.xx.xx.x' is local
0(2974) parse_headers: flags=16384
0(2974) pre_auth(): Credentials with given realm not found
0(2974) build_auth_hf(): 'Proxy-Authenticate: Digest realm="208.xxx.xxx.xx", nonce="43fe35e639b33d7c58a0d89e47df7fe11838584d"
'
0(2974) parse_headers: flags=-1
0(2974) check_via_address(208.xxx.xxx.xx, 208.xxx.xxx.xx, 0)
0(2974) DEBUG:destroy_avp_list: destroying list (nil)
0(2974) receive_msg: cleaning up
0(2974) SIP Request:
0(2974) method: <ACK>
0(2974) uri: <sip:8889997777@206.xx.xx.x>
0(2974) version: <SIP/2.0>
0(2974) parse_headers: flags=1
0(2974) Found param type 232, <branch> = <z9hG4bK40f0830c>; state=6
0(2974) Found param type 235, <rport> = <n/a>; state=17
0(2974) end of header reached, state=5
0(2974) parse_headers: Via found, flags=1
0(2974) parse_headers: this is the first via
0(2974) After parse_msg...
0(2974) preparing to run routing scripts...
0(2974) parse_headers: flags=4
0(2974) DEBUG: add_param: tag=2d38da4c38ea547b795405b32554c3bb.f3cb
0(2974) end of header reached, state=29
0(2974) DEBUG: get_hdr_field: <To> [73]; uri=[sip:8889997777@206.xx.xx.x]
0(2974) DEBUG: to body [<sip:8889997777@206.xx.xx.x>]
0(2974) DEBUG: sl_filter_ACK : local ACK found -> dropping it!
0(2974) DEBUG:destroy_avp_list: destroying list (nil)
0(2974) receive_msg: cleaning up
I am interested in a custom openser module. It needs to perform cnam
(calling party name) resolution through a third party webserver. The
module should perform the following actions on an incoming call where
SIP Display info: == " "(nothing).
1.Query our local cnam_custom db (mysql) to determine if the customer
being called has a custom cnam for the number calling them
2.Query our local cnam_cache db (mysql) to determine if we have
already queried the third party database for the number calling in
3.Query the third party database (https) to determine cnam if neither
of the first two queries returned anything
4.Insert the result into the SIP Display info: portion of the INVITE
so that the correct name will show up on my users callerid.
If 1,2,or 3 return a non null value skip to to 4.
I would really love to do this with an external script that grabs the
values, inserts them into the database, and AVP's but have been this
far been unable to build a working solution. I tried...
avp_db_load("$from","cnam/$cnam_scheme");
xlog("cnam value [$avp(s:cnam)]\n");
uac_replace_from("$cnam","");
or something like that to no avail. Please reply with your hourly
rate if you will need to write the module, if you can do it with
$avp's?, and include total cost for the job. I would prefer to do it
with avp's. Thank you.
Dear all,
1. How can I change username prefix 8 to other under
user registration ?
2. How can I set default timezone for all users ?
Best Regard,
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hi all,
does anyone know if SER supports Number Portability.
Thank you in advance
-----Messaggio originale-----
Da: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]Per conto di gokhan uslu
Inviato: venerdì 24 febbraio 2006 16.54
A: serusers(a)lists.iptel.org
Oggetto: [Serusers] radius authentication with ip address of a user
Hi,
I have installed ser and freeradius into same machine. everything is working fine. i have created users at the freeradius. but while creating a user, in the user information i had to use the ip of the SER not the ip of the user gateway(a sip gateway)
I mean i want to register the user according to its username , password and ALSO IP.
my user`s ip: 83.100.100.23
my SER ip: 212.175.10.36
when i edit user file for freeradius and when i write this:
myuser(a)83.100.100.23 for user name it does not work
but when i write
myuser(a)212.175.10.36
then it can authenticate. but this is unusefull for me at the moment. I must be authenticated according to user`s ip, not according to SER`s ip
please help meee
thanks a lot
_____
Relax. Yahoo! Mail virus <http://us.rd.yahoo.com/mail_us/taglines/virusall/*http://communications.yah…> scanning helps detect nasty viruses!
Hi,
I have installed ser and freeradius into same machine. everything is working fine. i have created users at the freeradius. but while creating a user, in the user information i had to use the ip of the SER not the ip of the user gateway(a sip gateway)
I mean i want to register the user according to its username , password and ALSO IP.
my user`s ip: 83.100.100.23
my SER ip: 212.175.10.36
when i edit user file for freeradius and when i write this:
myuser(a)83.100.100.23 for user name it does not work
but when i write
myuser(a)212.175.10.36
then it can authenticate. but this is unusefull for me at the moment. I must be authenticated according to user`s ip, not according to SER`s ip
please help meee
thanks a lot
---------------------------------
Relax. Yahoo! Mail virus scanning helps detect nasty viruses!
I'm doing an interop test with a well known
ITPS, they are kicking back these BYE retransmits, like:
SIP MESSAGE 18 216.138.115.30:5061(www.testnode-3.com) ->
209.247.16.2:5060(deneps551.Denver1)
UDP Frame 18 17/Feb/06 17:09:36.8142
TimeFromPreviousSipFrame=0.0010 TimeFromStart=8.0692
BYE sip:209.247.16.2:5060 SIP/2.0
Record-Route: <sip:216.138.115.30:5061;ftag=245-192.168.10.1;lr>
Record-Route: <sip:216.138.115.27;ftag=245-192.168.10.1;lr>
Via: SIP/2.0/UDP 216.138.115.30:5061;branch=z9hG4bK036d.a99621d6.0
Via: SIP/2.0/UDP 216.138.115.27;branch=z9hG4bK036d.1fe4ca41.0
Via: SIP/2.0/UDP 198.41.9.65:5060;branch=z9hG4bK245-192.168.10.1.256
Via: SIP/2.0/UDP 192.168.10.23:5060;branch=z9hG4bK1371-192.168.10.23.1374
From: <sip:+18663072489@216.138.115.30:5061>;tag=245-192.168.10.1
To: "Unknown" <sip:+17205626376@209.247.16.2>;tag=VPSF50603522629634
Call-ID: DEN05020060217230928060500(a)209.244.48.214
CSeq: 101 BYE
Max-Forwards: 67
Record-Route: <sip:198.41.9.65:5060>
Contact: "8663072489" <sip:+18663072489@198.41.9.65:1074>
SIP MESSAGE 19 216.138.115.30:5061(www.testnode-3.com) ->
209.247.16.2:5060(deneps551.Denver1)
UDP Frame 19 17/Feb/06 17:09:36.8424
TimeFromPreviousSipFrame=0.0282 TimeFromStart=8.0974
BYE sip:209.247.16.2:5060 SIP/2.0
Record-Route: <sip:216.138.115.30:5061;ftag=245-192.168.10.1;lr>
Record-Route: <sip:216.138.115.27;ftag=245-192.168.10.1;lr>
Via: SIP/2.0/UDP 216.138.115.30:5061;branch=z9hG4bK036d.a99621d6.0
Via: SIP/2.0/UDP 216.138.115.27;branch=z9hG4bK036d.1fe4ca41.0
Via: SIP/2.0/UDP 198.41.9.65:5060;branch=z9hG4bK245-192.168.10.1.256
Via: SIP/2.0/UDP 192.168.10.23:5060;branch=z9hG4bK1371-192.168.10.23.1374
From: <sip:+18663072489@216.138.115.30:5061>;tag=245-192.168.10.1
To: "Unknown" <sip:+17205626376@209.247.16.2>;tag=VPSF50603522629634
Call-ID: DEN05020060217230928060500(a)209.244.48.214
CSeq: 101 BYE
Max-Forwards: 67
Record-Route: <sip:198.41.9.65:5060>
Contact: "8663072489" <sip:+18663072489@198.41.9.65:1074>
Why is the BYE being sent so quickly?
This is an intermediate UAS proxy, and every now and
then it spits out an extra BYE like this. Shouldn't
this wait 500ms before doing a retry? The rest of the trace
is clean. I can forward the whole thing if there is any interest.
-g
--
Greg Fausak
greg(a)thursday.com
Hi Martin!
please always cc the list
Martin Petraschek wrote:
> I want openser to listen on port 5060 for UDP and TCP, and on port 5061
> for TCP/TLS. How could I achieve this? My current config is as follows:
>
> listen=udp:10.1.2.9:5060
> listen=tcp:10.1.2.9:5060
> disable_tls = 0
> listen = tls:10.1.2.9:5061
this is fine
> tls_port_no=5061
this is not needed
>
> Is this correct?
>
> > What is the request URI and destination before sending (try
> > xlog("L_ERR","duri=$du, ruri=$ru") before t_relay())
>
> With the openser.cfg as shown above the error message I described in my
> mail previous mail went away. However, when I try to make a call, I get
> the following:
>
> Feb 24 15:01:48 server2 /usr/local/sbin/openser[4246]: duri=,
> ruri=sip:user2@192.168.0.47:3985;transport=tls;line=ojn9itpa
> Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: child process
> 4246 exited by a signal 11
Sorry, i have no idea why it crashes here. Try to increase the debug
level (debug=4) and watch the logfile.
regards
klaus
> Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: core was generated
> Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: INFO: terminating
> due to SIGCHLD
>
> If you need more information, please tell me.
>
> Thanks,
>
> Martin
>
>
>
>
try Eyebeam, non free, but works great with ser:
http://www.xten.net/index.php
Ciao
--- Piergiorgio Venuti <piergiorgio(a)mediaservice.net> ha scritto:
> Hi all,
> who know any windows program that support multi video conferencing with
> ser, Windows Messanger work well with ser but not sopport this feature.
> Any idea?
>
> Tnx
>
> --
> +----------------------------------------------------------------------+
> | Venuti Piergiorgio Email: piergiorgio(a)mediaservice.net |
> | System Manager Tel: +39-011-32.72.100 |
> | @ Mediaservice.net S.R.L. Fax: +39-011-32.46.497 |
> | Via S.Bernardino 17 Torino |
> | Disclaimer: http://@Mediaservice.net/disclaimer |
> +----------------------------------------------------------------------+
>
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>
>
>
>
>
>
>
>
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