Hello
does openSER uses the host identificayion in any way to authenticate its
costumers?
Because I re-installed my server and thats the only difference from the
previous instalation (its even in the same machine).
Thanks
regards
Joao pereira
Hello,
with openser 1.0 I have this
if ( avp_db_load("$low_c_service/username","i:/serviceLowCoastredirect"))
and for openser 1.2 I try to do this
if (
avp_db_load("$avp(low_c_service)/username","i:/serviceLowCoastredirect"))
or
if (
avp_db_load("$avp($low_c_service)/username","i:/serviceLowCoastredirect"))
low_c_service is an alias
avp_aliases="low_c_service=i:500"
but I have always an error.
0(0) ERROR:avops:parse_avp_db:: error - bad avp flags
0(0) ERROR:avpops:fixup_db_avp: parse failed
0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 633
0(0) ERROR: fix_expr : fix_actions error
hi all;
i installed openser now trying to setup a basic text commnuication between two clients on the same pc (one client is windows msngr and the other one is x-lite )
it seems to me SUBSCRIBE and MESSAGE sip msgs are not processed.
i generated openser.cfg with the wizard at sipwise.com site
here is my ngrep dump for SUBSCRIBE msg
====================
U ua_public_ip:29350 -> 192.168.200.2:5060
SUBSCRIBE sip:burak@testsrv200 SIP/2.0.
Via: SIP/2.0/UDP ua_public_ip:29347.
Max-Forwards: 70.
From: "apo@testsrv200" <sip:apo@testsrv200>;tag=29774bb79657418eb518068a22304b7c;epid=13d631ff91.
To: <sip:burak@testsrv200>.
Call-ID: 15088cb5e87d4fb887cc073d0fd9df07.
CSeq: 1 SUBSCRIBE.
Contact: <sip:ua_public_ip:29347>.
User-Agent: RTC/1.3.
Event: presence.
Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml.
Supported: com.microsoft.autoextend.
Supported: ms-benotify.
Proxy-Require: ms-benotify.
Content-Length: 0.
.
#
U 192.168.200.2:5060 -> ua_public_ip:29350
SIP/2.0 501 Method Not Supported Here.
Via: SIP/2.0/UDP ua_public_ip:29347;rport=29350.
From: "apo@testsrv200" <sip:apo@testsrv200>;tag=29774bb79657418eb518068a22304b7c;epid=13d631ff91.
To: <sip:burak@testsrv200>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea73.
Call-ID: 15088cb5e87d4fb887cc073d0fd9df07.
CSeq: 1 SUBSCRIBE.
Server: OpenSER (1.2.0-notls (x86_64/linux)).
Content-Length: 0.
MESSAGE msg (while trying to establish a text chat)
===========================
U ua_public_ip:29355 -> 192.168.200.2:5060
MESSAGE sip:apo@testsrv200 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.33:58348;branch=z9hG4bK-d87543-2a3b75359734fe48-1--d87543-;rport.
Max-Forwards: 70.
To: "apo"<sip:apo@testsrv200>.
From: "burak"<sip:burak@testsrv200>;tag=fb1a9f68.
Call-ID: NTNhMGZlMmYwODIxNmFlNmNmYThmYjQ0NzFhOGFjODI..
CSeq: 2 MESSAGE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/im-iscomposing+xml.
User-Agent: X-Lite release 1006e stamp 34025.
Content-Length: 259.
.
<?xml version='1.0' encoding='UTF-8'?>
<isComposing xmlns='urn:ietf:params:xml:ns:im-iscomposing'
xmlns:xsi='http://www.w3.org/2001/XMLSchema-instance'>
<state>active</state>
<contenttype>goober</contenttype>
<refresh>60</refresh>
</isComposing>
#
U 192.168.200.2:5060 -> ua_public_ip:29355
SIP/2.0 501 Method Not Supported Here.
Via: SIP/2.0/UDP 192.168.1.33:58348;branch=z9hG4bK-d87543-2a3b75359734fe48-1--d87543-;rport=29355;received=ua_public_ip.
To: "apo"<sip:apo@testsrv200>;tag=329cfeaa6ded039da25ff8cbb8668bd2.402d.
From: "burak"<sip:burak@testsrv200>;tag=fb1a9f68.
Call-ID: NTNhMGZlMmYwODIxNmFlNmNmYThmYjQ0NzFhOGFjODI..
CSeq: 2 MESSAGE.
Server: OpenSER (1.2.0-notls (x86_64/linux)).
Content-Length: 0.
.
.
Hi list.
I am having a problem with siptrace module. When I send the command
sip_trace on through fifo, openser do not send back responses to user
agents.
On logfile (with debug=9), I just receive this message:
sip_trace: storing info...
On database do not appear any trace message.
Thanks in advance for your attention.
Kind regards.
Sergio G.
Hi,
I am using openser 1.1.x as a front end to asterisk.
Here is what happens in the dialog:
1. phone sends invite
2. openser says "Unauthorized"
3. phone sends invite + Authorization header
4. openser auths the packet and forwards to asterisk
5. asterisk says "proxy authentication required"
6. openser says "proxy authentication required"
here is where it gets interesting.
7.a On some devices, the phone sends invite + Authorization header +
Proxy-Authorizaiton header.
7.b On some devices, the phone sends invite + Proxy-Authorization header.
How does one work around 7.b?
Really, what I want to see is how to make it so that asterisk does not
try to authenticate the call. However, since that doesn't seem to be
possible, what is the sensible way to route these calls?
Thanks,
Mark Price
I'm trying to setup Openser (1.1) as registrar with mediaproxy and mysql all
on the same box.
I am currently using the SIP Proxy/Registrar with Mediaproxy config from
sipwise. MySQL and Mediaproxy are running. Xlite registers successfully with
Openser with the users I have created. The problem comes with the
authentication. Both www_authorize and proxy_authorize fail. www_authorize
fails at regtistration and proxy_authorize off course then fails when I try
make a call.
Any suggestions?
Thanx
Hi,
In openser 1.1.x, is it possible to get similar functionality to that
suggested by the following:
avp_check($hdr(ua[0]),"re/Some Phone UA/")
thanks
Mark Price
Hi openser,
Following the tutorial on FreeRadius integration (http://www.openser.org/docs/openser-radius-1.0.x.html), I'm seeing accounting packets successfully sent to Radiusd. However, some attributes are missing, notably: UserName, CallingStationId, and CalledStationId.
Radius client dictionary contains:
ATTRIBUTE User-Name 1 string
ATTRIBUTE Password 2 string
...
ATTRIBUTE Called-Station-Id 30 string
ATTRIBUTE Calling-Station-Id 31 string
Output from 'radiusd -X' shows:
...
Module: Instantiated detail (detail)
Listening on authentication *:1812
Listening on accounting *:1813
Ready to process requests.
rad_recv: Accounting-Request packet from host 10.10.90.198:32812, id=10, length=120
Acct-Status-Type = Start
Service-Type = IAPP-Register
Attr-102 = 0x000000c8
Error-Cause = 1
Event-Timestamp = "Apr 4 2007 15:10:27 PDT"
Attr-105 = 0x3937333153495070546167303031
Attr-104 = 0x3937323853495070546167303131
Acct-Session-Id = "1-9731(a)10.10.90.51"
NAS-Port = 5060
Acct-Delay-Time = 0
NAS-IP-Address = 10.10.90.198
Processing the preacct section of radiusd.conf
modcall: entering group preacct for request 0
modcall[preacct]: module "preprocess" returns noop for request 0
rlm_acct_unique: WARNING: Attribute User-Name was not found in request, unique ID MAY be inconsistent
...
Any tips on how to fix this? Thanks.
I think the key message for problem analysis, client's ACK, is missing. This is the setting,
right:?
.100/UAC -----NAT/.248----> SER outbound @ .246 -----> UAS @ .239
The ACK in your message dump is only that one sent from proxy (246) to UAS
(.239). The ACK which can possibly mismatch and produce the error message
you mentioned must come from upstream. To be properly formatted, it must
have Via identical to that of initial INVITE
( Via: SIP/2.0/UDP 192.168.1.100:5060;rport=46127;received=100.100.100.248;branch=z9hG4bK2893084087)
-jiri
At 17:50 27/03/2007, Ricardo Martinez wrote:
No. Time Source Destination Protocol Info
> 14 6.102719 100.100.100.239 100.100.100.246 SIP Status: 487 Request Terminated
>Session Initiation Protocol
> Status-Line: SIP/2.0 487 Request Terminated
> Status-Code: 487
> Resent Packet: False
> Message Header
> Via: SIP/2.0/UDP 100.100.100.246;branch=z9hG4bK895c.a98d35.0
> Via: SIP/2.0/UDP 192.168.1.100:5060;rport=46127;received=100.100.100.248;branch=z9hG4bK2893084087
>No. Time Source Destination Protocol Info
> 15 6.103011 100.100.100.246 100.100.100.239 SIP Request: ACK sip:005622408196@100.100.100.239
>Session Initiation Protocol
> Request-Line: ACK sip:005622408196@100.100.100.239 SIP/2.0
> Method: ACK
> Resent Packet: False
> Message Header
> Via: SIP/2.0/UDP 100.100.100.246;branch=z9hG4bK895c.a98d35.0
> f: "cl" <sip:cl@sipvoiss.desa.redvoiss.net>;tag=1587516403
> SIP Display info: "cl"
> SIP from address: sip:cl@sipvoiss.desa.redvoiss.net
> SIP tag: 1587516403
> i: <mailto:e94a98aa177e8579e5242a2091c0ac5f@192.168.1.100>e94a98aa177e8579e5242a2091c0ac5f(a)192.168.1.100
> To: <sip:0299005622408196@sipvoiss.desa.redvoiss.net>;tag=9146c297a4
>Thanks
>
>Ricardo.-
>_______________________________________________
>Serusers mailing list
>Serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hi all,
I am trying to implement call redirection feature using cpl script.
I have complied cpl module that supports the cpl script.
I have uploaded the cpl script.
plz help me to solve the problem..
I have written this mail regarding the error that
> we are getting in testing the cpl script of Call redirection.
>
> We have three X-lite phones with SIP uris 2001(a)172.16.1.24,
> ,2002(a)172.16.1.28 and 2003(a)172.16.1.31 and all the three soft phones
> have been registered with SER proxy running on 10.2.3.123. All the
> three users have added in the SER database along with mail ids.
>
> user mail id
> 2001 2001(a)172.16.1.24
> 2002 2002(a)172.16.1.28
> 2003 2003(a)172.16.1.31
>
> The call scenario is as follows..
> 1.user 2001 will make a call to user 2002.
> 2. If the user 2002 is either not responding (No answer) or busy, then
> the call
> will be redirected to user 2003.
> 3. we able to make a call to 2002,and user 2002's X-lite phone is
> ringing but the call is
> not getting redirected.
>
> The cpl script for redirection has been deployed in SER proxy.
>
> I am sending the call log of sip phone:2001 as an attachment along
> with snapshot sip phone 2001.
>
> I am sending cpl script at the bottam of this mail.plz find the cpl script
at the end of this mail.
> As shown in the xlite screen "500 CPL script execution failed" can u
> please suggest me if there is
> any error in XML/CPL syntax.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> ------------------------------------------------------------------------
>
> ------------------------------------------------------------------------
>
> <?xml version="1.0"?>
> <!DOCTYPE html cpl PUBLIC "-//IETF//DTD RFCxxx CPL 1.0//EN" "cpl.dtd">
> <cpl>
> <subaction id="pda">
> <location url="sip:2003@172.16.1.31 <sip:1003@172.16.1.31>">
> <proxy/>
> </location>
> </subaction>
> <incoming>
> <location url="sip:2002@172.16.1.28 <sip:1002@172.16.1.28>">
> <proxy timeout="8">
> <busy>
> <sub ref="pda"/>
> </busy>
> <noanswer>
> <sub ref="pda"/>
> </noanswer>
> </proxy>
> </location>
> </incoming>
> </cpl>
