Deal all,
I am a newbie in SIP
I am trying to add Voicemail service on my SER .
As according to "SIP Express Router v0.11 --Admin Guide" SEMS is required to provide voicemail capability in SIP server. i have downloaded "sems-0.1.0.tar.gz" and installed in my machine where I have "ser-0.8.11" as a sip server.
I have changed my ser.cfg scripts to provide voicemail capabilities and to load vm module available in SER.
Now I get stuck How I have to use SEMS module ( I mean to say that "ans_machine" available in sems )
what is the difference & similarity between SEMS Voicemail and SER vm module.
On what basis SEMS Voicemail and SER communicate to each other.
Is their any configuration file is required inside the SEMS like SER (ex. ser.cfg).
what is the proper step to perform a testing of voice mail using these modules..
(the same problem witk koyama)
Thanks for your help....
Erdem H. HAKI
I'm testing SER with Windows Messenger 4.7 and I have problems with
the presence service. Those contacts connected when the user connects,
appear as connected (correct). But when another contact connects, it
keeps appearing as disconnected, and after a long time (more or less
one minute) shows the contact as connected.
On the other hand, all the contacts appear as disconnected with
Windows Messenger 5.1, independently that the contact is previously
connected or connects later.
Has someone problems with windows messenger?. I've checked that these
contacts are connected in SER with "serctl ul show". So I imagine that
it's some type of incompatibility with windows messenger.
Hi!
Can anyone tell me which Hardware requirements I need
for a SER server ?
Is there anywhere a documentation for this ?
Regards,
Ahmad
--
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service(a)NetUSE.DE -- http://NetUSE.DE/
-----Oorspronkelijk bericht-----
Van: Paul van Schagen [mailto:paul.van.schagen@adaline.net]
Verzonden: donderdag 21 april 2005 9:02
Aan: 'Josekutty K.K'
Onderwerp: RE: [Sipforum-discussion] regarding Authentication...
Methods you really want to authenticate are INVITE ( outgoing calls )
and REGISTER ( incoming calls ). You try the request first and the
switch answers requesting authentication. Then the authentication is
sent by the UA to the switch.
Remember the authentication also verifies the IP address of the sender
is correct. By doing a double request you make sure that the original
request came from where they say it came from.
Lets say somebody fakes an IP address and sents the packet to the
switch.
The switch then sends back an authentication requests which arrives at
another destination. The call never gets authenticated correctly and is
rejected.
Hopes this helps
Can anybody help me with my mediaproxy timeout ?
Best regards
Paul van Schagen
-----Oorspronkelijk bericht-----
Van: sipforum-discussion-bounces(a)lists.su.se
[mailto:sipforum-discussion-bounces@lists.su.se] Namens Josekutty K.K
Verzonden: donderdag 21 april 2005 8:47
Aan: sipforum-discussion(a)lists.su.se
Onderwerp: [Sipforum-discussion] regarding Authentication...
Hi all,
I am a beginner in SIP. Let me explain my doubt.
Once a SIP request authenticates itself, (usually a REGISTER request),
all further requests MUST include the authorization details .. I mean
I MUST send the authorization details to subsequent requests say
OPTIONS, INVITE etc . Or Should I try each request without
authentication first?
Regards
Josekutty K K
------
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Hey All,
Do any of you know of a reliable / knowledgeable source for commercial
technical support for SER.
If so, we may be interested.
We have worked in the past directly with IPTel, and have discussed support
options recently with Jiri. He suggested that we ask the fine folks of the
SER Users mailing list for their suggestions.
Thanks so much,
Darren Nay
Ionosphere, Inc
dnay(a)ionosphere.net
(864) 678-3158
hi
how to get call duration in radius. there should be
callduration in accounting stop but here in my case
there is no callduration
Acct-Status-Type = Stop
Service-Type = Sip-Session
Sip-Response-Code = 200
Sip-Method = 8
User-Name = "4000@ser"
Calling-Station-Id = "sip:4000@ser"
Called-Station-Id = "sip:3000@ser"
Sip-Translated-Request-URI = "sip:3000@ser"
Acct-Session-Id =
"CB13EC9C-9DB4-420F-BEF4-786E6398D714@UA"
Sip-To-Tag = "as23df87de"
Sip-From-Tag = "290015013442"
Sip-Cseq = "2"
NAS-IP-Address = 127.0.0.1
NAS-Port = 5060
Acct-Delay-Time = 0
Client-IP-Address = 127.0.0.1
Acct-Unique-Session-Id = "ca12eca5fce40009"
Timestamp = 1112912023
__________________________________________________
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Hi Steve!
Now it works properly. Like you wrote the port command
was missed in the dial peer.
dial-peer voice 1 pots
description Default-Dial-peer fuer ausgehende Anrufe
preference 3
service session
max-conn 25
destination-pattern 0T
progress_ind alert enable 8
direct-inward-dial
port 0/0/1:15
Now I have activated the www_authorize section.
# Uncomment this if you want to use digest authentication
if (!www_authorize("netuse.de", "subscriber")) {
www_challenge("netuse.de", "1");
break;
};
On my cisco I see following when I enable the ccsip debugging:
Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 192.168.254.30:5060
Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.203:5060;branch=z9hG4bK2924EA
From: "0*" <sip:0*@192.168.254.30>;tag=E7AF75C-13AE
To: "0*" <sip:0*@192.168.254.30>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2283
Call-ID: E1A23104-B0DA11D9-81B582BA-75F29624
CSeq: 42 REGISTER
WWW-Authenticate: Digest realm="netuse.de", nonce="426756705993d4a9785e2e828158f76faa5234d4", qop="auth"
Server: Sip EXpress router (0.9.0 (sparc64/solaris))
Content-Length: 0
Warning: 392 192.168.254.30:5060 "Noisy feedback tells: pid=16087 req_src_ip=192.168.254.203 req_src_port=52773 in_uri=sip:192.168.254.30:5060 out_uri=sip:192.168.254.30:5060 via_cnt==1"
Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x45D7009C key=E1A23104-B0DA11D9-81B582BA-75F29624
Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 45D7009C
It looks like the cisco tries to registering to the server, but the
server decline this.
How can I enable the registering of my gateway in ser, without to disable
the www_authorize ?
Regards,
Ahmad
On Apr 20, 05, Steve Blair wrote:
>
>
> Ahmad Cheikh Moussa wrote:
>
> >Hi!
> >
> >>
> >>>Can I ask you some questions?
> >>>How do I know, that my ser server only act as registrar and
> >>>redirect Server ? With redirect I mean, any other calls, which
> >>>starts with 0, redirect to the gateway.
> >>
> >>
> >>
> >>I'm not sure I understand. Are you concerned about people outside
> >>of your user community routing calls through your proxy and out to the
> >>PSTN? If so then you need to use check_from and check_to to
> >>determine if the caller is authenticated to your proxy.
> >
> >I have following line in ser.cfg
> >t_relay_to_udp("192.168.254.203", "5060");
> >when I understand you right, then I have to change this paramater to
> >sl_relay_to_udp("192.168.254.203", "5060");
> >
> No, the t_relay is fine.
>
> >This would mean stateless forwarding to this gateway ?
> >correct ?
> >
> >Regars,
> > Ahmad
> >
>
> --
>
> ISC Network Engineering
> The University of Pennsylvania
> 3401 Walnut Street, Suite 221A
> Philadelphia, PA 19104
>
>
> voice: 215-573-8396
>
> 215-746-8001
>
> fax: 215-898-9348
>
> sip:blairs@upenn.edu
>
--
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service(a)NetUSE.DE -- http://NetUSE.DE/
Hi!
I have some problems with SER 0.8.14. When I'm trying AUTH modules, it crashes. (it says:
---------------------------------------------------------------------------------------------------
Starting SER : cat: /var/run/ser.pid: No such file or directory
started pid()
---------------------------------------------------------------------------------------------------
My config is the following:
#
# $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
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"
# 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 (len_gt( 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("ignosya.dyndns.org", "subscriber")) {
www_challenge("ignosya.dyndns.org", "0");
break;
};
save("location");
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();
};
}
Hi,
I loaded serweb on my gentoo box and when I open the my_account.php I
get a error highlighted in red sorry - fifo reading error.
Has anyone else encountered this problem and if so how can I fix it.
Wercs Communications
Clay Bryan
Network Administrator
Wercs Communications
400 East First
Casper, WY 82601
<http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe
r%2C+WY++82601&country=us>
CBryan(a)wercs.com <mailto:CBryan@wercs.com>
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mobile:
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307-233-8701
307-258-7371
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