At 02:11 AM 2/10/2004, kapil dhawan wrote:
>thats absolutely fine i know md5 is used....i can also compare coming password in requested with my md5 converted password but i am stuck with how and where to read pasword information coming in a request
The authentication module does it for you. It takes the MD5-hashed value
from request (example bellow) and compares it against expected value based
on credentials stored in subscriber table. The table may be located in mysql,
dbtext or whereever.
-jiri
example:
Authorization: DIGEST username="17479383213", realm="proxy01.foobar.com", algorithm=MD5, uri="sip:proxy01.foobar.com", nonce="4028321e2577ba09167c3c5702bc2105d560dbaa", response="6e71e5747df3198d24d00fb2a8733392"
>>From: Jiri Kuthan <jiri(a)iptel.org>
>>To: "kapil dhawan" <oswriter(a)hotmail.com>, serusers(a)lists.iptel.org
>>Subject: Re: [Serusers] Password
>>Date: Mon, 09 Feb 2004 08:27:26 +0100
>>
>>At 05:04 AM 2/9/2004, kapil dhawan wrote:
>>>Hi
>>>How can i retrieve password coming in 'REGISTER' Request.
>>
>>There is fortunately no easy way to do it. SIP used MD5 digest
>>function which would be quite hard to reverse for you.
>>
>>>problem is i don't want to do mysql authentication as i have less space...
>>
>>use dbtext module.
>>
>>-jiri
>
>_________________________________________________________________
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--
Jiri Kuthan http://iptel.org/~jiri/
Hello.
I have succeeded in operation of CPL function.
However, there is a script which does not operate in the following cases.
<Non-answered transmission by CPL>
incoming-> location url -> proxy timeout="10"-> noanswer-> proxy --
You need to point out the problem of my script
----------------------------------------------------
<?xml version="1.0"?>
<!DOCTYPE cpl PUBLIC "-//IETF//DTD RFCxxxx CPL 1.0//EN" "cpl-06.dtd">
<cpl>
<incoming>
<location url="sip:1000@192.168.100.11:5060">
<proxy timeout="10">
<noanswer>
<location url="sip:2000@192.168.100.11:5060">
<location url="sip:3000@192.168.100.11:5060">
<proxy ordering="parallel" />
</location>
</location>
</noanswer>
</proxy>
</location>
</incoming>
</cpl>
------------------------------------------------------
Taniguchi / PanasonicCC
Hi,
Attached with this email is the ethereal o/p for communication that
happens between two wince clients registering on ser using TCP. My problem
is that once conversation starts between the two clients within moments a
network error occurs and the converstaion stops. I am trying to trace the
SIP communication between the server and the client to see if there is any
clue as to why the communication terminates all of a sudden. Experts please
help me with your insight on this.
Thanks,
Annie
I'm interested in using the append_rpid_hf function.
The auth module doc states that the function inserts
"the saved value of the SIP URI received from the
database or radius server followed by . . . "
I've inserted the function into my config file, plus
carefully inserted the necessary loadmodule and
modparam statements but no RPID field is being created
[based on ethereal captures at the gateway]. I think
that its because I'm not doing the right thing to get
a SIP URI saved to memory. I'm not sure what I'm
missing here.
Also, my subscriber table has no RPID column in it so
I pointed the rpid fuction to the phone column.
Also, will the radius module work with any [i.e. non
linux/unix] radius server?
I'm running 0.8.12 on RH 9
version: ser 0.8.12-tcp_nonb (i386/linux)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168.4.1 2003/11/26 23:45:50
andrei Exp $
main.c compiled on 12:58:28 Jan 2 2004 with gcc 3.2
Thanks,
G.
__________________________________
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thx we got it in action.c and implemented it also..
Now there is one more issue i like to discuss......
what we are doing is...have written a module and every INVITE request we are
forwarding it to that place......
checks done there
1. is call allowed
2. is call denied
3. has call to be forwarded...etc and few more....
Now if we are forwarding it to some number so in ser.cfg we just relay the
call after writing new uri into the structure.....now if we want to do the
same checks of allow , deny and forward etc on the same call....will
recursion do it in our module or we have to play with ser.cfg.....
regards
>From: Jan Janak <jan(a)iptel.org>
>To: kapil dhawan <oswriter(a)hotmail.com>
>CC: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] Rewrite URI
>Date: Wed, 11 Feb 2004 20:19:34 +0100
>
>See rewrite_RURI function in sip_router/modules/rr/loose.c
>
> Jan.
>
>On 11-02 09:27, kapil dhawan wrote:
> > I want to rewrite URI in one of my modules. How can i do it....i know to
>do
> > it in ser.cfg file but not in my module.....
> >
> > _________________________________________________________________
> > Easiest Money Transfer to India. Send Money To 6000 Indian Towns.
> > http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
_________________________________________________________________
Masterpieces made affordable! Buy art prints.
http://go.msnserver.com/IN/42736.asp MSN Shopping.
I am battling this RADIUS Authentication configuration for SER. Does anyone
have a SER.CFG file they can send me of a working RADIUS setup configuration
file
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
Vonage locks the VoIP configuration menus on the VT1000v units they sell.
Does anyone know where to get these units other than from Vonage or how to
unlock the VoIP config menus?
-----Original Message-----
From: Alan Crosswell [mailto:alan@columbia.edu]
Sent: Wednesday, February 11, 2004 5:47 PM
To: Andy Singh
Cc: 'serusers(a)lists.iptel.org'
Subject: Re: [Serusers] Motorola VT1000v with Ser
I guess Vonage switched from the Cisco ATA-186.
Andy Singh wrote:
> Just wondering if any tried using Motorola VT1000v box with Ser. It's a
SIP
> UA and you can plug in your analog phone into it to make calls. You can
find
> out more on this at
>
> http://broadband.motorola.com/consumers/products/vt1000v/ and company
> called vonage sells the VOIP service using this to consumer.
>
> Thanks
> Andy
>
> _______________________________________________
> 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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* Arnd Vehling <av(a)nethead.de> [040211 22:26]:
> Youre sende address isnt valid. i..e camro instead of camaro
Thanks.. Just one of my typos..
Ill talk my boss, and see if they are intrested in sponsoring the
software.. Was'nt there a guy who said he where gonna do it a month or
2 ago ? or do I remeber wrong?
- Atle
>
> -------- Original Message --------
> Subject: failure notice
> Date: 11 Feb 2004 20:21:43 -0000
> From: MAILER-DAEMON(a)ns.voipnet.net
> To: av(a)nethead.de
>
> Hi. This is the qmail-send program at ns.voipnet.net.
> I'm afraid I wasn't able to deliver your message to the following addresses.
> This is a permanent error; I've given up. Sorry it didn't work out.
>
> <clona(a)camro.no>:
> Sorry, I couldn't find any host named camro.no. (#5.1.2)
>
> --- Below this line is a copy of the message.
>
> Atle Samuelsen wrote:
> >He he.. Saw the NDA from sipquest.. Did'nt want to sign it myself..
>
> Yes, quite annoying. Especially when considering that the product
> is based on a former open source software (called siph323).
>
> Unluckily the siquest guys have been very successfull in removing
> all open source copies from the net. At least i wasnt able to
> find any.
>
> Right now i am talking to a guy who offered enhancing Vovidas
> siph323csgw so it can bridge between SIP and H323 networks.
> I am looking for other persons who are willing to financially
> support this project. The costs would be between EU 500 and EU 300
> and the result would be returned to the open source community.
> Details of this would need to be worked out though.
>
> Contact me if youre interested in this.
>
> -- Arnd
>
>
Hi List,
We have a few local domains that we will be using SER act as a proxy and to
protect our gateway. In the near future we will also be providing PSTN
breakout for other SIP domains that are not on our proxy.
What is the best way to allow traffic from certain know SIP domains out
through our gateways?
I've thought about having users authenticate to our proxy, but this could
get very complex and messy serving multiple local & remote domains. Other
option is just to allow traffic through from the trusted domains I guess.
One last question, is any one familiar with the Cisco AS5300?
How do you go about locking it down so that only authorised users can pass
calls out onto the PSTN?
We have a number of Vega 100's that have the option of only allow proxy
invited calls. Is there something similar on the AS5300 or would that have
to do authentication aswell, AAA ?
I know that this question is slightly off topic, but with it being related I
thought someone my be able to answer my question.
Many Thanks,
Alan
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