hi,
First we signed in a(a)localhost.localdomain and which will subscribe b(a)localhost.localdomain's presence info.
Yes, we do receive 200 OK for that SUBSCRIBE. After that SER keep sending NOTIFY to a(a)localhost.localdomin with status "closed", even after b(a)localhost.localdomain signed in, and b also get same staff.
Can you help us ? the ser.cfg is attached below.
thanks,
hong
-------------------
What exactly is not working ? Do you get any 200 OK for the SUBSCRIBE
followed by a NOTIFY ?
Jan.
On 04-12 11:41, hong cheng wrote:
> hi all,
>
> Try to use presence feature with MS messager 5.0, seems SUBSCRIBER can not work.
> Can anybody have a look for the attached ser.cfg ?
>
> Many thanks,
> hong
>
> ----------------------------------
>
> #debug=4 # debug level (cmd line: -dddddddddd)
> #listen=192.168.8.96
> #fork=no
> #log_stderror=yes # (cmd line: -E)
>
> #debug=7
> #fork=no
> #log_stderror=yes
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
> port=5060
> children=4
> fifo="/tmp/ser_fifo"
>
> # ------------------ module loading ----------------------------------
>
> # Uncomment this if you want to use SQL database
> #loadmodule "/usr/local/lib/ser/modules/mysql.so"
>
> loadmodule "/usr/local/lib/ser/modules/sl.so"
> loadmodule "/usr/local/lib/ser/modules/tm.so"
> loadmodule "/usr/local/lib/ser/modules/rr.so"
> loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
> loadmodule "/usr/local/lib/ser/modules/usrloc.so"
> loadmodule "/usr/local/lib/ser/modules/registrar.so"
> loadmodule "/usr/local/lib/ser/modules/pa.so"
>
>
>
> # Uncomment this if you want digest authentication
> # mysql.so must be loaded !
> #loadmodule "/usr/local/lib/ser/modules/auth.so"
> #loadmodule "/usr/local/lib/ser/modules/auth_db.so"
> # ----------------- setting module-specific parameters ---------------
>
> # -- usrloc params --
>
> modparam("usrloc", "db_mode", 0)
>
> # Uncomment this if you want to use SQL database
> # for persistent storage and comment the previous line
> #modparam("usrloc", "db_mode", 2)
>
> # -- auth params --
> # Uncomment if you are using auth module
> #
> #modparam("auth_db", "calculate_ha1", yes)
> #
> # If you set "calculate_ha1" parameter to yes (which true in this config),
> # uncomment also the following parameter)
> #
> #modparam("auth_db", "password_column", "password")
>
> # -- rr params --
> # add value to ;lr param to make some broken UAs happy
> modparam("rr", "enable_full_lr", 1)
> # ------------------------- request routing logic -------------------
>
> # main routing logic
>
>
> route{
>
> # initial sanity checks -- messages with
> # max_forwards==0, or excessively long requests
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> break;
> };
> if ( msg:len > max_len ) {
> sl_send_reply("513", "Message too big");
> break;
> };
>
> # we record-route all messages -- to make sure that
> # subsequent messages will go through our proxy; that's
> # particularly good if upstream and downstream entities
> # use different transport protocol
> record_route();
> # loose-route processing
> if (loose_route()) {
> t_relay();
> break;
> };
>
> # if the request is for other domain use UsrLoc
> # (in case, it does not work, use the following command
> # with proper names and addresses in it)
> if (uri==myself) {
>
> if (method=="REGISTER") {
>
> # Uncomment this if you want to use digest authentication
> # if (!www_authorize("iptel.org", "subscriber")) {
> # www_challenge("iptel.org", "0");
> # break;
> # };
>
> save("location");
> break;
> };
>
> if (method=="SUBSCRIBE") {
> if (t_newtran()) {
> handle_subscription("registrar");
> };
> break;
> };
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> break;
> };
> };
> # forward to current uri now; use stateful forwarding; that
> # works reliably even if we forward from TCP to UDP
> if (!t_relay()) {
> sl_reply_error();
> };
>
> }
Jan,
Thank you for the response.
The longer I use the SER proxy server, the more uses I
find for it.
What I am currently envisioning is this:
I run a PBX PROXY server for a specific business
customer. This customer has a private naming (numbering)
plan. The names of the UA devices are 101, 102, 103, etc..
These are 'extension numbers'.
I would like to be able to make the PBX 'pretend' it is
the phone. It will send a REGISTER to the GATEWAY, and then subsequent
INVITES for that number will be sent to the PBX. The PBX will
determine how to deliver the call using local rules on the
SER proxy. The rules will be something like:
business days between 9am and 5pm forward all incoming calls to x100.
If extension 100 is busy (or doesn't answer), roll to extension 101.
If extension 101 is busy (or doesn't answer), roll to voice mail.
And a million other things.
I realize that I can forward the call from the GATEWAY
to the PBX, but, I think it would be MUCH cleaner if
I could tell the SER PBX engine to REGISTER a number....
Make sense?
---greg
>
>Greg,
>
>it would be possible with some shell script magic. It is possible to
>send a REGISTER message using the fifo interface.
>
>Before we dig into into it, let me ask what is the reason for doing
>something like that ? Can't you simply forward the SIP messages from
>GATEWAY to PBX and if PBX is off or crashed then the GATEWAY will send
>"Request timeout" to the PSTN (instead of 404) ? Or am I missing
>something ?
>
> Jan.
>
>On 03-12 21:59, Greg Fausak wrote:
>> Hello,
>>
>> I have a SER server, it has 2 phones
>> registered to it. The phones are 300000
>> and 300100. The registration works fine, and
>> I can place calls between them. I am calling this
>> my 'PBX'.
>>
>> I have another SER server that can connect to
>> a PSTN. I can register my phone number with it,
>> and I can make and receive calls from the PSTN.
>> Let's say that phone number is 2145551212(a)addaline.com.
>> I'm calling this my 'GATEWAY'.
>>
>> OK, here is the tricky part. I would like to make
>> the SER 'PBX' *REGISTER* 2145551212(a)addaline.com with
>> the SER GATEWAY ... just like
>> it is a UA. So an incoming call to 2145551212 would
>> get delivered to the SER PBX, and I'll figure out which
>> extension gets the call with local rules. For very
>> simplistic delivery at the PBX I can alias 2145551212 as
>> 300000.
>>
>> If the PBX crashes, or is turned off, the 2145551212
>> registration will eventually evaporate.
>>
>> Anyway, I can't figure out anyway to get SERPBX to register
>> with SER GATEWAY.
>>
>> Is this possible?
>>
>> ---greg
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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
>
Hi!
> -----Original Message-----
> From: Martin Anderberg [mailto:martin@namnupplysaren.net]
> Sent: Thursday, December 04, 2003 5:26 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] symmetric nat/ broadband routers
>
>
>
> Howdy!
>
> I'm having some trouble with the nathelper module and certain
> types of
> broadband routers (ie d-link 604 & d-link 624). I'll try to
> explain the
> situation below and hope that someone is willing to help me
> out, because
> i'm stuck.
>
> In short, the setup is a ata-186 box (which is symmetric) behind a
> d-link 604 (which isn't symmetric at all times).
>
> The nathelper module included in the distribution (both 0.8.11 and
> 0.8.12) has a function called fix_nated_contact().
> fix_nated_contact()
> rewrites the contact-header with the source-ip & source-port
> of the packet.
>
> However, in some cases (ie non-symmetric d-link router between the
> ata-box and the internet) this is a problem since the d-link router
> sometimes rewrites the source-port which is then used as a
> location in
> ser. When the session has timed out on the d-link (doesn't
> really seem
> to help with the natping) the location-information in ser is
> no longer
> valid.
>
> Is there any reason why the nathelper rewrites the port in the
> contact-header? If the client is symmetric the source-port
> and the port
> in the contact-header shouldn't differ anyway? I trust there is a
> reason, i just dont see it ;)
>
Yes, the ports at the client are identical, but the NAT router uses
other ports at the public interface, e.g. if the client behind the NAT
uses port 5060 for SIP (send&receive), the NATs publice interface may
user for example port 50000. Therefore, the client listen on port 5060,
but the packets have to be sent to port 50000 of the public IP address
and then the NAT router rewrites the port back to 5060. Hence, the
nathelper modules rewrites the IP address and the port in the contact
header before saving them in the location database.
If the session in the NAT router times out although using natping, thats
a pit. Maybe it helps to ping the proxy from the client, e.g. the
budgetone phones support keep alive pinging.
Klaus
> On a side-note, when glancing at nathelper.c it looks as if
> the int len
> is calculated with the original values of the header, then filled
> through snprintf with values which are not 100% positively the same
> length (msg->rcv.src_port). Isn't it for example possible
> that the port
> in the header is 5060 but the source_port is 22444 (which is one
> character longer than the length of len is calculated to).
>
> I hope someone can shed some light over the matter.
>
> /Martin
>
>
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
When I uncomment this in my ser.cfg:
if (!www_authorize("telepacket.com", "subscriber")) {
www_challenge("telepacket.com", "0");
break;
};
my Cisco 7960 phones are unable to register with the ser Server.
Basically, the phone sends a REGISTER, and the ser server sends back 401 Unauthorized. How do I authorize the phones? If I comment these lines, the ser works great and my two phones can call each other.
I am using a mysql database... Is there a command to add a registered device?
NGREP shows this:
U 206.194.77.243:50208 -> 206.194.77.241:5060
REGISTER sip:206.194.77.241 SIP/2.0..Via: SIP/2.0/UDP 206.194.77.243:5060..From: sip:ctest@206.194.77.241..To: sip:ctest@206.194.77.241..Call-ID: 00036b3c-2e33007e-653539fe-
314cbc28@206.194.77.243..Date: Thu, 04 Dec 2003 21:40:40 GMT..CSeq: 102 REGISTER..User-Agent: CSCO/4..Contact: sip:ctest@206.194.77.243:5060..Authorization: Digest username=
"LINE 2",realm="telepacket.com",uri="sip:206.194.77.241",response="1c90541dc3d33153cfd9d278c2171e18",nonce="3fcfaa82021d76bd83d8ef539ff0a49ca92e2d54",algorithm=md5..Content-
Length: 0..Expires: 3600....
#
U 206.194.77.241:5060 -> 206.194.77.243:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 206.194.77.243:5060..From: sip:ctest@206.194.77.241..To: sip:ctest@206.194.77.241;tag=b27e1a1d33761e85846fc98f5f3a7e58.904a..Call-
ID: 00036b3c-2e33007e-653539fe-314cbc28@206.194.77.243..CSeq: 102 REGISTER..WWW-Authenticate: Digest realm="telepacket.com", nonce="3fcfaa82021d76bd83d8ef539ff0a49ca92e2d54"
..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 206.194.77.241:5060 "Noisy feedback tells: pid=13375 req_src_ip=206.194.77.243 req_src_p
ort=50208 in_uri=sip:206.194.77.241 out_uri=sip:206.194.77.241 via_cnt==1"....
Hi,
We are in the process of doing extensive testing with the new SIPURA SPA2000
unit on our network. (SER 0.8.11)
One question has come up regarding the time on the unit itself. It does not
support NTP but the developers say that if we can insert a Date header in a
REGISTER response then the unit can pick up the time from that. This is an
exact quote from them:
" We thought that it would be
simpler to deploy if the proxy server (or Gateway, or other
entity) can insert a Date header (with GMT time)..
.
.
Is it easy for
you to insert a Date header in a REGISTER response, for
instance?
"
Seems like a practical idea but I am unsure exacly how to go about doing
this. Any suggestions?
Thanks,
Andres.
Hi!
sems version 1.0 is for ser0.8.11
the cvs version of sems is for ser0.8.12
klaus
> -----Original Message-----
> From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
> Sent: Friday, November 28, 2003 5:50 PM
> To: serusers
> Subject: [Serusers] Re: voicemail config
>
>
> Hi,
>
> thanks for your help! I have now the two instances of ser working ok.
> The vm ser instance also writes the message to the fifo and this is
> picked up by Sems. However, Sems is complaining about the "wrong FIFO
> interface version". Sems seems to pick up version 0.2. This
> matches what
> the ser proxy writes to the FIFO. I run ser 0.8.12 (stable
> version) and
> sems version 0.1.0 which appears to be the most recent.
>
> How can I make ser and sems agree on the FIFO interface version?
>
> Thanks,
> Mario
>
>
>
> --
> The University of Stirling is a university established in Scotland by
> charter at Stirling, FK9 4LA. Privileged/Confidential Information may
> be contained in this message. If you are not the addressee indicated
> in this message (or responsible for delivery of the message to such
> person), you may not disclose, copy or deliver this message to anyone
> and any action taken or omitted to be taken in reliance on it, is
> prohibited and may be unlawful. In such case, you should destroy this
> message and kindly notify the sender by reply email. Please advise
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>
> _______________________________________________
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> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hi all !
My ser.cfg (ser-0.8.12) is very similar like nathelper.cfg
I run rtpproxy as ./rtpproxy -l IPaddress.
On this same interface is ser binded
Communication between ser and rtpproxy is via /var/run/rtpproxy.sock
Why I'm not able to connect with rtpproxy ?
I have:
Dec 4 19:48:45 /usr/sbin/ser[19200]: ERROR: extract_mediaip: no
`c=' in SDP
Dec 4 19:48:45 /usr/sbin/ser[19200]: ERROR: send_rtpp_command: can't connect to RTP proxy
Dec 4 19:48:45 /usr/sbin/ser[19200]: ERROR: extract_mediaip: no `c=' in SDP
Dec 4 19:48:45 /usr/sbin/ser[19200]: ERROR: send_rtpp_command: can't connect to RTP proxy
Dec 4 19:48:45 /usr/sbin/ser[19208]: WARNING: check_content_type:
Content-TYPE header absent!let's assume the content is text/plain ;-)
Dec 4 19:48:45 /usr/sbin/ser[19208]: ERROR: on_reply processing failed
Regards
Andrzej
Hi everybody!
I want to install ser in my own Linux account (I am not root). Is there
any way to do that?
Thx!
Karim.
--
---------------------------
Karim SBATA
INT - Dpt LOR
9 rue Charles Fourier
91011 Evry
tel: +33 1 60 76 44 43
fax: +33 1 60 76 47 11
web: http://www.sbata.fr.st
---------------------------
Shouldn't 'commercial grade' firewalls support SIP/RTP? I know cisco
firewalls doing SIP and NAT traversal very well.
Klaus
> -----Original Message-----
> From: Hans Eriksson [mailto:hansa@mac.com]
> Sent: Thursday, December 04, 2003 6:13 PM
> To: Klaus Darilion
> Cc: <serusers(a)lists.iptel.org>
> Subject: Re: [Serusers] symmetric nat/ broadband routers
>
>
> Klaus,
>
> Many commersial grade firewalls do not keep sessions alive,
> regardsless
> of external pings, so it won't work in rather too many cases.
>
> Also, assuming many users (10k, 100k) doing natpings will be heavy.
>
> But alas, NATs are a hack and maybe the only remedies will also be
> hacks, with all the pros and cons.
>
> cheers
> /hans
>
> 4 dec 2003 kl. 17.43 skrev Klaus Darilion:
>
> > Yes, the ports at the client are identical, but the NAT router uses
> > other ports at the public interface, e.g. if the client
> behind the NAT
> > uses port 5060 for SIP (send&receive), the NATs publice
> interface may
> > user for example port 50000. Therefore, the client listen
> on port 5060,
> > but the packets have to be sent to port 50000 of the public
> IP address
> > and then the NAT router rewrites the port back to 5060. Hence, the
> > nathelper modules rewrites the IP address and the port in
> the contact
> > header before saving them in the location database.
> >
> > If the session in the NAT router times out although using natping,
> > thats
> > a pit. Maybe it helps to ping the proxy from the client, e.g. the
> > budgetone phones support keep alive pinging.
> >
> > Klaus
>
>
Hello List,
Has anyone tried upgrading their Grandstream Budgetone firmware?
I downloaded the latest firmware from Grandstream support website - but the tftp client on the phone is looking for some additional files (cfg.txt, vp.bin and bt.bin) that are not included in the firmware. I have written to Grandstream support but haven't heard anything from them. Anyone has any answer for this? I am using Budgetone 100 SIP Phone.
Thanks,
Dhiraj Bhuyan
Network Security Specialist,
BT Exact Business Assurance Solutions
Tel: +44 1473 643932
Mob: +44 7962 012145
Email: dhiraj.2.bhuyan(a)bt.com
