Dear all,
i have problems in using SERWEB. i get the latest serweb in berlios. i'm running ser on redhat 9. my version ser is 8.12. mysql 3.x.x , sendmail 8.12 and php 4.x.x. my ser and my serweb are running well but when someone want to subscribe via serweb the e-mail confirmation doesn't arrive to user who subscribe and serweb say: register is successfully. i have check php.ini, httpd.conf, and sendmail configuration, they all seems OK. i tried to solve this problems in a week but the result is ZERO.
please help me .
Regards,
Sita, Politeknik Caltex Riau
Indonesia
Sita Rosita
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dear all,
im using ser 8.12,redhat 9.0,php.4.x,mysql.3.x,sendmail.8.x. and serweb i download the latest version on berlios.
all of my ser service is running ok, but when some one access serweb to subscribe as new users. in my serweb it's say okay and you are sucessfully and the email confirmation will be sent shortly. but when i check in users email, the email isn't arrived. how comes. i just try this in a intranet.
are anyone have idea and hint???
regard's
sita (politeknik caltex riau)
indonesia
Sita Rosita
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Great.
One thing I was thinking is that if most people need this value to be set smaller then someone
should commit a change to CVS this this at 8 to 10 rather than 20.
Thanks,
Paul
--- Richard <richard(a)o-matrix.org> wrote:
> Hi Java,
>
> That value is to prevent double detection, i.e. the same key is detected
> twice. Too large, it runs the risk of not detecting the right key. To small,
> it increases the chance of double detection.
>
> I use 7 or 8 in my system.
>
> Richard
>
> > -----Original Message-----
> > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
> > Behalf Of Java Rockx
> > Sent: Thursday, November 04, 2004 5:44 PM
> > To: Java Rockx; ser users
> > Subject: Re: [Serusers] sems IVR plugin bug???
> >
> > Hi All.
> >
> > Perhaps I just fixed this, but will my change cause other problems? The
> > change I made was in
> > IvrDtmfDetector.h
> >
> > I changed
> >
> > DTMF_INTERVAL = 20
> >
> > to
> >
> > DTMF_INTERVAL = 10
> >
> > And now dialing DTMF digits is ok. Will reducing DTMF_INTERVAL to 10 cause
> > other issues?
> >
> > Regards,
> > Paul
> >
> > --- Java Rockx <javarockx(a)yahoo.com> wrote:
> >
> > > Hello All.
> > >
> > > I'm using sems which I updated from berlios CVS tonight (2004-11-04).
> > >
> > > There appears to be a bug in the DTMF detection in the IVR plugin.
> > >
> > > Using the DTMF-detection.pl test script that ships with sems, DTMF
> > detection works fine for
> > > non-repeating digits. However, it always drops a digit if you repeat a
> > number.
> > >
> > > For example:
> > >
> > > GOOD ===> Press 123456 and the IVR plugin will receive 123456
> > > BAD ===> Press 001122 the IVR plugin will receive 012
> > >
> > > I've tested this with my grandstream BT100 using DTMF mode as
> "in-audio",
> > "via RTP", and "via
> > > SIP
> > > INFO". The "in-audio" and "via RTP" both produce this problem while the
> > "via SIP INFO" digits
> > > are
> > > never detected (which is understandable).
> > >
> > > I searched the mailing list but found no reference to this problem.
> > >
> > > So the question is this; Does anyone have a patch to fix this?
> > >
> > > Regards,
> > > Paul
> > >
> > >
> > >
> > >
> > >
> > > __________________________________
> > > Do you Yahoo!?
> > > Check out the new Yahoo! Front Page.
> > > www.yahoo.com
> > >
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> >
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > Check out the new Yahoo! Front Page.
> > www.yahoo.com
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
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Greetings,
I have been reading a lot about SER lately, trying to figure out the
intricacies, but cannot find out how to achieve one thing. I want to be
able to get SER to register to another SIP Proxy(another SER?) to
receive a given prefix. I'm trying to achieve a combination of Asterisk
and SER, but the box I have SER running on must also be connected to
another Proxy which is not under my control. I read that SER is a SIP
Proxy which can also act as a registrar. It would seem logical that It
is possible to get my SER box to register to the other SIP Proxy to tell
it to forward the given prefix to me so that I can pass it off
elsewhere. But how do I achieve this? Is it possible? All help is
greatly appreciated.
Thank you.
Tom Gaudasinski
hi everyone :
there are some problems about ser server's jabber module and database
location.
1.when i configure the jabber module.there are some errors about database
sip_jab.However I cannot find the database sip_jab in the server.The
configuration is
modparam("jabber", "db_url", "sql://ser:helso@127.0.0.1/sip_jab)
2.there are two tables in databse ser:"server_monitoring" and
"server_monitoring_agg".But when the server is running.There is no datas in
the tables.Do the monitor infomations write into them?
3.In the database ser,there is a table location.However when a user log
in.The field "domain" is still null.How can I write the domain infomation
into the table "location"?
How can i solve the problem?
These are some informations related to my setting and problem:
1、operating system:Linux 7.0
2、SER distribution: ser-0.8.14_linux_i386.tar.gz
3、SER build: version: 0.8.14 (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 1.168.4.3 2004/06/28 15:41:21 andrei Exp $
main.c compiled on 12:28:01 Jul 27 2004 with gcc 2.95
4、SER configuration file :
debug=7 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
check_via=yes # (cmd. line: -v)
dns=yes # (cmd. line: -r)
rev_dns=yes # (cmd. line: -R)
port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="voipv6.edu.cn" "210.25.130.252" "localhost"
# ------------------ 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"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/print.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/pa.so"
loadmodule "/usr/local/lib/ser/modules/jabber.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
# ----------------- setting module-specific parameters ---------------
modparam("usrloc","db_url","sql://ser:heslo@localhost/ser")
# -- 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)
modparam("usrloc", "use_domain", 1)
# -- auth params --
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)
modparam("jabber", "db_url", "sql://ser:helso@voipv6.edu.cn/sip_jab)
modparam("jabber", "jaddress", "voipv6.edu.cn")
modparam("jabber", "jport", 5222)
modparam("jabber", "jdomain", "voipv6.edu.cn=*")
modparam("jabber", "aliases", "1;msn.x.com=%")
# ------------------------- 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=~"voipv6.edu.cn") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("voipv6.edu.cn", "subscriber")) {
www_challenge("voipv6.edu.cn", "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();
};
}
_________________________________________________________________
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I sent this earlier and got no responses. Perhaps this is not the right
forum to ask this question. Can any one suggest a better place to go for
this information?
Thanks, Bob
-----Original Message-----
From: Bob Carlson
Sent: Wednesday, November 03, 2004 3:22 PM
To: 'SerUsers'
Subject: Transfer and Conferencing
Let me apologize in advance for my question, which is a little rudimentary.
We are just starting a project that will use SER and I am being forced to
document right now how transfer and conferencing will be handled. I have
spent a lot of time looking for definitive information on the subject with
no luck. Well, maybe too much luck. There seem to be many proposals and
models and so on, but it is not clear to me what is actually being done in
practice. I have downloaded all the RFCs and proposal papers on the
subject. I am still reviewing them, but I think the folks on this forum can
help me a lot.
I need to know the SIP message sequences for performing a call transfer and
a blind call transfer and for constructing a conference. I have found
information in proposals, but I need to know what actual, available SIP
phones can do. We have some phones that we will test, but I do not know
what they do when you press their transfer and conference buttons. Pardon
me again for my impatience in asking before I have tried this out.
The Transfer models are straightforward, but conferencing is more
complicated. We must construct a simple conferencing model where the
conferencing is performed by a central server, a SIP IPX. Only conferences
of 3 participants need to be supported. We want it to look exactly like
3-way calling on your home phone. During a call, put the call on hold with
a conference button, call another phone, hit conference button, the two
calls are joined in a 3-way conference.
The document draft-ietf-sipping-service-examples-07.txt seems to be very
helpful on the subject, but all examples are in the form of 3 or more UAs
and do not address any examples from the point of view of a PBX. I can see
how to extend the examples to a PBX case, except for one aspect. If the
IP-PBX is to perform the action as a proxy, what does the phone send the
IP-PBX to indicate the steps in the process. Put more plainly, what happens
when the user hits the Transfer or Conference button on the phone? What
message is sent to the IP-PBX?
Can anyone tell me where else I should be looking? Is the service examples
draft the best base document to work from?
Thanks in advance, Bob Carlson
Hello All.
I'm using sems which I updated from berlios CVS tonight (2004-11-04).
There appears to be a bug in the DTMF detection in the IVR plugin.
Using the DTMF-detection.pl test script that ships with sems, DTMF detection works fine for
non-repeating digits. However, it always drops a digit if you repeat a number.
For example:
GOOD ===> Press 123456 and the IVR plugin will receive 123456
BAD ===> Press 001122 the IVR plugin will receive 012
I've tested this with my grandstream BT100 using DTMF mode as "in-audio", "via RTP", and "via SIP
INFO". The "in-audio" and "via RTP" both produce this problem while the "via SIP INFO" digits are
never detected (which is understandable).
I searched the mailing list but found no reference to this problem.
So the question is this; Does anyone have a patch to fix this?
Regards,
Paul
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Check out the new Yahoo! Front Page.
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Flynn
modules being loaded properly.
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Starting simple switch on 'Zap/1-1'
Nov 5 09:06:43 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)...
Nov 5 09:06:45 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)...
Nov 5 09:06:46 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)...
-- Executing Goto("Zap/1-1", "default|12345|1") in new stack
-- Goto (default,12345,1)
-- Executing Ringing("Zap/1-1", "") in new stack
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing AGI("Zap/1-1", "xml.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/xml.agi
-- Playing 'agent-pass' (language 'en')
-- xml.agi: The Input data is #,queuename is group2
-- xml.agi: User Input : #
-- xml.agi: get in GetAgent group2
-- xml.agi: no agent is avail
-- xml.agi: no agent
-- Started music on hold, class 'default', on Zap/1-1
-- xml.agi: get in GetAgent group2
-- xml.agi: no agent is avail
-- xml.agi: get in GetAgent group2
-- xml.agi: no agent is avail
-- xml.agi: get in GetAgent group2
-- xml.agi: no agent is avail
-- xml.agi: get in GetAgent group2
-- xml.agi: no agent is avail
........................................
--soft hangup zap/1-1
Requested Hangup on channel 'Zap/1-1'
-- Stopped music on hold on Zap/1-1
== Spawn extension (default, 12345, 3) exited non-zero on 'Zap/1-1'
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
// I have found from google
it is common problem
http://www.marko.net/asterisk/archives/0208/0329.html
This has been covered in many other messages, but in order for the X100P
to detect hangup, you must have disconnect supervision on the phone line.
If the line you are providing is from another PBX, then it is highly
unlikely that it supplies the disconnect supervision. Most telephone
switches do support disconnect supervision, but it's not always on by
default. You can tell by using a lighted keypad which receives its power
only from the phone line, and then calling it and hanging up on it. If
the lighted keypad blinks off for a moment then your line has the
disconnect supervision, otherwise it doesn't.
http://www.marko.net/asterisk/archives/0206/0198.html
"ks" is the right one. Have you confirmed that your line supports
disconnect supervision? Here is an easy test:
Get a phone that has a lighted keypad, where the lighted keypad is powered
ONLY by the phone line's power. Call the phone and then hang up on it.
If the lighted keypad goes out for a brief time, then you do have
disconnect supervision and we have to figure out how to tune the driver to
see it.
>why don't you try using the stable version of asterisk? right now i'm
>using version 1.0RC2 (which is the version just before the actual
>1.0.0) and it's been fantastic.
>
>What you're describing could be a whole bunch of things -- IRQ sharing
>problems, faulty configuration files, etc. Why don't you attach a bit
>more information, perhaps the zapata.conf file.
>
>are the appropriate modules being loaded properly? Try watching the
>output of "dmesg" to see if there are any problems with the digium
>card.
>
>Flynn
>
>
>On Thu, 4 Nov 2004 10:08:34 +0800, dev2003 <dev2003(a)mail.ustc.edu.cn> wrote:
>> Flynn,
>>
>> this sort of behaviour happenning all the time.
>> such as when call 11,then I hangup.
>> but when I recall 11,then it is busy.
>> os redhat 9.
>> /usr/sbin/asterisk -r
>> Asterisk CVS-HEAD-09/10/04-21:34:12, Copyright (C) 1999-2004 Digium.
>> Written by Mark Spencer <markster(a)digium.com>
>>
>> zaptel-0.9.0
>>
>>
>>
>>
>> >can you give more details about your asterisk settings? when is this
>> >sort of behaviour happenning -- all the time? or only when a specific
>> >condition occurs?
>> >
>> >flynn
>> >
>> >p/s including your config files wouldn't be a bad idea as well
>> >
>> >
>> >On Wed, 3 Nov 2004 18:33:28 +0800, dev2003 <dev2003(a)mail.ustc.edu.cn> wrote:
>> >> serusers，您好！
>> >>
>> >> asterisk can not hangup .user Wildcard X100P.
>> >> when using phone call,asterisk can not hangup.
>> >> I should tpye:
>> >> soft hangup zap/1-1
>> >> then can hangup.
>> >>
>> >> dev2003
>> >> dev2003(a)mail.ustc.edu.cn
>> >> 2004-11-03
>> >>
>> >> _______________________________________________
>> >> Serusers mailing list
>> >> Serusers(a)iptel.org
>> >> http://mail.iptel.org/mailman/listinfo/serusers
>> >>
>> >>
>> >>
>>
>> = = = = = = = = = = = = = = = = = = = =
>>
>> 致
>> 礼！
>>
>>
>> dev2003
>> dev2003(a)mail.ustc.edu.cn
>> 2004-11-04
>>
>>
>>
= = = = = = = = = = = = = = = = = = = =
致
礼！
dev2003
dev2003(a)mail.ustc.edu.cn
2004-11-05
FYI: I tried to run seweb with PHP 5.0.2. It sort of works, but at
least one function that serweb relies upon has changed in v5:
Warning: array_merge() [function.array-merge]: Argument #1 is not an
array in /home/sipa/var/serweb/html/user_interface/my_account.php on
line 286
More info on:
http://www.php.net/manual/en/function.array-merge.php
I switched to php 4.3 and all is well.
John
