Hi
Most versions of windows messenger are only for XP. Check http://www.meetingbywire.com/VersionWatch.htm
They have pretty complete overview of the different messengers from MS. I heard once that MSN Messenger 5 Beta still had once SIP implementation. Or try also one of the older versions which support your OS.
Good luck
Ralph
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Rao
Sent: Montag, 29. März 2004 22:33
To: Jan Janak
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] problem with IM b/w windows messenger 4.7 and 5.0
Jan, Ralph and Daniel thanks for your responses.
Here is some more calrification.
1) I am using windows messaeger 5.0 and MSN messenger.
2) The link Ralph provided only works for wondows XP.
On one of my machines I am running windows 2000
and would like to see IM work on win2K. I cannot
find IM 4.7 for any version of windows except XP.
3) I can not get IM to work even when both
machines are running messenger 5.0. Seems to be
the same issue when one is running 4.7 i.e.
maddr in route header is being ignored and ser
tries to deliver it to the local machine and
tcp connect fails.
Has any one made IM to work b/w MM 5.0
Following is a trace of a SIP message request that
has maddr in the Route request. Let me know if more
traces are needed.
So I have following questions
1) Can there be an maddr parameter in the Route
header ? Cannot find an answer in RFC 3261.
2) In any case why can't we fix ser to interpret
maddr.
windows OS is probably the most important OS
with which ser needs to support.
3) What is a better utility than ngrep to get traces
which are better formatted. I have seen folks post
very nicely formatted traces, what are they using.
Let me know if I need to post these questions
on the development list.
Regards.
Rao.
##
T 129.150.32.27:1064 -> 129.146.175.207:5060 [AP]
MESSAGE sip:rao-toshiba@129.146.175.207;
ftag=3b3b0bff72424892aa8e78d5bbf3a316;lr=on
SIP/2.0..Via: SIP/2.0/TCP 129.150.32.27:7685..
From: <sip:rao-toshiba@sipserver>;
tag=6d5db818-3a2a-40b3-b919-5cd06b18d8e9..
To: "rao-sony@sipserver" <sip:rao-sony@sipserver>;
tag=3b3b0bff72424892aa8e78d5bbf3a316;
epid=8bb54d4c1b..
Call-ID:
72452f0ccee6437193a271e057d52725(a)129.146.85.163..
CSeq: 4
MESSAGE..Route:<sip:rao-sony@sipserver:13395;
maddr=129.146.85.163;transport=tcp>..
Contact: <sip:129.150.32.27:7685;transport=tcp>..'
User-Agent: Windows RTC/1.0..Content-Type:
text/plain;
charset=UTF-8;msgr=WAAtAE0ATQBTAC0ASQBNAC0ARgBvAHIAbQBhAHQAOgAgAEYATgA9AE0AUwA
lADIAMABTAGgAZQBsAGwAJQAyADAARABsAGcAOwAgAEUARgA9ADsAIABDAE8APQAwADsAIABDAFMAP
QAwADsAIABQAEYAPQAwAA0ACgANAAoA..Content-Length:
20....sending from toshiba
#
T 129.146.175.207:5060 -> 129.150.32.27:1064 [AP]
SIP/2.0 477 Unfortunately error on sending to next
hop occured (477/TM)..
Via: SIP/2.0/TCP 129.150.32.27:7685..
From: <sip:rao-toshiba@sipserver>;
tag=6d5db818-3a2a-40b3-b919-5cd06b18d8e9..
To: "rao-sony@sipserver" <sip:rao-sony@sipserver>;
tag=3b3b0bff72424892aa8e78d5bbf3a316;epid=8bb54d4c1b..
Call-ID:
72452f0ccee6437193a271e057d52725(a)129.146.85.163..
CSeq: 4 MESSAGE..Server: Sip EXpress router (0.8.12
(sparc64/solaris))..
Content-Length: 0....
#
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Serusers mailing list
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At 08:00 PM 3/29/2004, Jev wrote:
>Because the phones that are being used are in common areas, used by employees, they work like traditional phones (using ata186's) so to use the phone they just pick it up and dial, these users don't log in.
However the ATAs still do authenticate using preconfigured password. -jiri
>I think this makes sense :)
>
>-Jev
>
>Jiri Kuthan wrote:
>
>>why dont you use digest authetnication to determine user identity? -jiri
>>At 08:40 AM 3/29/2004, Jev wrote:
>>
>>>Hi all,
>>>
>>>I wish to block outgoing calls to certain numbers, specifically mobile phone numbers, and international numbers. Thats pretty straight forward.
>>>
>>>However, when a user attempts to ring one of these numbers I would like to have a IVR system to prompt them for a PIN number, upon entering a valid pin number the user would be forwarded to the dialled number, and the call logged.
>>>
>>>Has any one put together a setup similar to this?
>>>
>>>Thanks,
>>>-Jev
>>>
>>>_______________________________________________
>>>Serusers mailing list
>>>serusers(a)lists.iptel.org
>>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>>--
>>Jiri Kuthan http://iptel.org/~jiri/
--
Jiri Kuthan http://iptel.org/~jiri/
Hello All,
What is the best SIP Client used with SER ?
Can SJphone be used with SER ? If yes, how to set up SJphone for authentification ?
Thank you very much for any info.
Regards,
Anton
checking from against digest credentials takes first verifying the
digest credentials with proxy_authenticate().
Note that this works for single domain. You can't really authenticate
a BYE if the party that hangs up is from some other adminsitrative
domain.
-jiri
At 11:17 PM 3/29/2004, Ticknor.Scott(a)ic.gc.ca wrote:
>my lab partner & i have found that if we sniff an ACK message during call
>setup and extract the call tag and id, then we can arbitrarily hang up the
>call from our java attack generator. i thought about adding some logic to
>ser.cfg to process BYEs. is there an easy way to authenticate the BYE? i
>have something like the following in ser.cfg, but it seems to have no effect
>
>if (method=="BYE") {
> if (!check_from()) {
> ...etc
> };
>};
>
>thanks,
>scott
>DSi
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
What client do you use? It does not use the realm from challenge -- it
is a MSN Messenger specific bug, but it might be present in other sip
clients. You must set the realm from challenge as the host part of the
sip id.
.Daniel
On 03/26/04 15:46, Raymond Chen wrote:
>Danial,
>
>www_authorize fail to authorize without password
>
>0(3283) lookup(): '85234230599(a)218.20.229.53' Not found in usrloc
> 0(3283) parse_headers: flags=4096
> 0(3283) pre_auth(): Credentials with given realm not found
> 0(3283) build_auth_hf(): 'WWW-Authenticate: Digest realm="xxx.org",
>nonce="40644192d74bf39b0ebb5d141cb2073a6c09daf8"
>'
>
>Regards
>
>Raymond
>
>-----Original Message-----
>From: Daniel-Constantin Mierla [mailto:daniel@iptel.org]
>Sent: Friday, March 26, 2004 8:04 PM
>To: Raymond Chen
>Subject: Re: [Serusers] RE: [Serdev] check_from
>
>
>Try something like this:
>
>if (method=="INVITE")
>{
> if (!www_authorize("xxx.org", "subscriber")) {
> www_challenge("xxx.org", "0");
> break;
> };
> if (!check_from()) {
> sl_send_reply("403", "Only registered users are allowed");
> break;
> };
>
>};
>
>.Daniel
>
>On 03/26/04 12:48, Raymond Chen wrote:
>
>
>
>>Now we understand what the what the message means after reading the message
>>a few times. We are trying to do PSTN(as5300) ---> ser -----> pstn
>>(AS5300), and to authorize the calling number (callerid) in the INVITE
>>message against URI table. But check_from command needs to call
>>proxy_authorize, which it requires username and password. we setup the
>>configuration like this
>>
>>if (method=="INVITE" & proxy_authorize("xxx.org", "subscriber")
>> if (!check_from()) {
>> sl_send_reply("403", "Only registered users are allowed");
>> break;
>> }
>>}
>>
>>Because cisco does not have sip password setting, so we have
>>
>>0(3173) check_username(): No authorized credentials found (error in
>>
>>
>scripts)
>
>
>>0(3173) check_username(): Call {www,proxy}_authorize before calling
>>
>>
>check_*
>
>
>>function !
>>
>>Does anyone has a solution?
>>
>>Regards
>>
>>
>>-----Original Message-----
>>From: Daniel-Constantin Mierla [mailto:daniel@iptel.org]
>>Sent: Friday, March 26, 2004 6:18 PM
>>To: Raymond Chen
>>Cc: serdev(a)lists.iptel.org
>>Subject: Re: [Serdev] check_from
>>
>>Hello,
>>the last error message is self explanatory. You need to call either
>>www_authorize() or proxy_authorize() before calling check_from() because
>>this method compares the data from From header with what is in
>>credentials (response to a authentication challenge).
>>
>>.Daniel
>>
>>On 03/26/04 04:35, Raymond Chen wrote:
>>
>>
>>
>>
>>
>>>Dear all,
>>>
>>>We have configured Ser to check from username field to authorize user
>>>“unknown”
>>>
>>>if (!check_from()) {
>>>
>>>sl_send_reply("403", "Only registered users are allowed");
>>>
>>>break;
>>>
>>>};
>>>
>>>We have error message
>>>
>>>0(2568) check_username(): No authorized credentials found (error in
>>>scripts)
>>>
>>>0(2568) check_username(): Call {www,proxy}_authorize before calling
>>>check_* function !
>>>
>>>We have “unknown” username entry in uri table.
>>>
>>>Regards
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>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
>>
>>
>>
>>
>>
>
>_______________________________________________
>Serdev mailing list
>serdev(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serdev
>
>
>
Hi,
I'm trying to integrate small ISDN networks into a SIP network, so quite
the opposite of the usual way (SIP -> PSTN).
There exist some SOHO ISDN-to-SIP Gateways with two BRI out there, and
I've successfully tested some of them by using the user id of the
gateway as MSN on the phone.
Now this is quite a waste of ISDN, cause you can in fact address up to
eight devices on an S0 Bus. My current solution is to use the alias
table of SER to configure the MSNs for the Gateway. The Gateway itself
registers with e.a. sip:011000@my.domain and there are aliases from
0110001 to 0110008 pointing to that URI.
But this still does not satisfy me because I think the usual way would
be to configure one base number which the ISDN2SIP-Gateway uses to
register with my SER, and let the user assign as much phone extensions
as he likes without any interaction of mine.
Here I'm thinking about a loop in the ser config where I lookup the
location of the user part of the To-Header (if "^[0-9]+$"), and if not
found, just cut off the last digit and repeat the loop until the user id
under-runs a minimum length. If no appropriate user is found, I return
404. Otherwise I forward the original URI to the IP of the user I've found.
So, generally said, I assign base numbers and perform some kind of
wildcard matching on incoming URIs.
What do you think about this solution? Would this be appropriate, or do
there exist other approaches?
I'm looking forward for some comments,
Andy
Hi, I´ll like to have runung several sip domains into 1 SER box, this is because I need to make for diferente sip users, diferent dial plan , (or tell me if there is a more easy way to do the following:
I want so each domain has their dial plan for example userA1 dial 150 to call userD1
sipdomain1.com
userA1
userB1
userC1
userD1
sipdomain2.com
userA2
userB2
userC2
userD2
userE2
Sorry for the inconvenience I´m a newbie I´ll apreciate explcit instructions as possible.
Regards
HA