Hi guys,
i met the problem with using serctl to add the new users. I have the FC4 on my pc.
I have uncommented the ser.cfg file for using mysql. and the error information is like follow:
[cluo@pc5 ~]$ /usr/sbin/serctl add richard 000000 richard_luo(a)lycos.com
/usr/sbin/serctl: line 764: gen_ha1: command not found
HA1 calculation failed
And if i changed to the root user, the error information is changed like these:
/usr/sbin/serctl: line 401: mysql: command not found
could anyone help me for this problems?
best
richard
============================
You won't admit you love me, and so How am I ever to know You always tell me...
Perhaps, perhaps, perhaps
--
_______________________________________________
Search for businesses by name, location, or phone number. -Lycos Yellow Pages
http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…
I've the following configuration :
internet --- ser -||-- asterisk
where ser has two network interfaces, one with public ip and one with
192.168.1.1, asterisk sits in the 192.168.1.0 net,
authorization and forwarding of calls from internet works well with this
ser.cfg (I omit parameters and obvious stuff):
if (uri==myself) {
if (method=="REGISTER") {
if (!radius_www_authorize("")) {
www_challenge("", "0");
break;
};
save("location");
break;
};
lookup("aliases");
};
if (method=="INVITE") {
record_route();
};
rewritehost("192.168.1.101"); <--- asterisk ip
if (!t_relay()) {
sl_reply_error();
};
but of course you can't hear nothing with this config...
Googling I only find configurations relative to the ser machine also
being natted, can someone tell me what I need to do in my particular case ?
Hi All,
Is my query correct
SELECT t1.from_uri as orig_number, t1.to_uri as term_number,
TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate
FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND
((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR
(t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND
t1.sip_method='INVITE' AND
t2.sip_method='BYE'
also this query displays all calls fron all user and it shows also all the
previos month, what can I add here to show only calls from a certain user
e.g 2009 and the current month?
I tried this:
SELECT t1.from_uri as orig_number, t1.to_uri as term_number,
TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate
FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND
((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR
(t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND
t1.sip_method='INVITE' AND t2.sip_method='BYE'
AND t1.username="2009" and t1.time REGEXP '-08-'; <--- I added this line
but i had an error, hope someone can help me. thank you.
regards,
nhadie
Hi everybody,
Based on the latest dialog supported (which was added in RR module), I
was able to implement in the "uac" module proper FROM replacement. The
missing issue was correct restoring/replacement of FROM in the
sequential requests. As restoring information was kept in FROM param,
this information didn't show up in all sequential requests (for example
in ones from original caller to callee).
Now, based on the new RR API, the module keeps the restoring info in the
Route header which will show up for sure in all sequential request; so
the module will be able to track and fix all sequential replies and
requests. This implies fixing FROM or TO hdr, depending of the request
direction.
For people not interested in implementation details, I put first the
usage description:
There are 3 restore mode for the module to work (for FROM mangling):
"none" -> NO RESTORE; the FROM URI will be changed and no further
effort or info will be done/store in order to restore its value neither
is sequential replies, nor requests;
"manual" -> information needed for restoring the value will stored
in RR parameter; all the replies and local generated requests (ACK and
CANCEL) will be automatically fixed, but no sequential request will be
restored; the admin has full liberty to do it from script by using the
restore_from() function.
"auto" -> FULL AUTO RESTORING -> information needed for restoring
the value will stored in RR parameter and all sequential request and
replies will be fixed *automatically*
NOTE: for restore modes different than "none", the module will try to
bind itself to RR and TM modules, so take care and load them.
For people interested in implementation details.
Shortly, here is how it works:
Let's consider a call initiated from UA A to UA B (or from A to B)
Now, since sequential requests may flow from A to B or B to A we need to
be able both to restore original TO in requests from B to A and to
replace with new FROM in request from A to B; which means that during
the dialog we need to know both old and new FROM uris. As in each
situation on of the value is known (depending of the direction, either
the new, either the new URI appears), the simplest and most efficient
way to store both in minimum size is to store the XOR value of them;
having the XOR value and one of the uris, we can determine the other URI
( (x^y)^y=x :) )
A Proxy
B
1) initial request:
From:OLD_URI FROM:NEW_URI
RR:(OLD_URI^NEW_URI)
--------------------------------->
----------------------------------->
2) final reply
From:OLD_URI FROM:NEW_URI
RR:(OLD_URI^NEW_URI) RR:(OLD_URI^NEW_URI)
<---------------------------------
<-----------------------------------
3) re-invite (direction downstream)
From:OLD_URI FROM:NEW_URI
Route:(OLD_URI^NEW_URI)
--------------------------------->
----------------------------------->
4) final reply
From:OLD_URI FROM:NEW_URI
<---------------------------------
<-----------------------------------
5) BYE (direction upstream)
TO:OLD_URI TO:NEW_URI
Route:(OLD_URI^NEW_URI)
<---------------------------------
<-----------------------------------
6) final reply
TO:OLD_URI TO:NEW_URI
--------------------------------->
----------------------------------->
1) initial equest has the FROM URI changed from scrip via replace_from()
; XOR between NEW_URI and OLD_URI is added to RR as parameter
2) the reply (actualy all replies) is fixed via TM callbacks
3) the re-invite is detected via RR callback as having the RR param; as
the direction is downstream,it will have the OLD_URI *FROM* changed with
the NEW_URI = (OLD_URI^NEW_URI)^OLD_URI
2) the reply is fixed via TM callbacks
3) the BYE is detected via RR callback as having the RR param; as the
direction is upstream,it will have the NEW_URI *TO* changed with the
OLD_URI = (OLD_URI^NEW_URI)^NEW_URI
and this is all the magic behind :)
regards,
bogdan
Hi
I cant recall the name of the proggie to help relay ser_fifo from one
machine to another, has anyone installed it, and does it work properly,
I want to move fifo away from the central install of ser
iqbal
Hello,
Any use of SDP media attribute in conjunction with SIP /SER ?
I would appreciate any insight!
George
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hi all...
Just came across a problem with running 2 individual SER for 2 separate network.
They both work fine independently, yet I have subscribed for PSTN landing of
calls for only 1 SER server (IP restrictions). Now, for my second SER server, I
have forwarded calls to the first server (much like pointing a call to it using
"if uri=~" command........ and of course, in the trusted table of SER 1, I
have placed the IP of the SER 2. Now... the call goes through and the can be
landed, yet the PSTN user can not hear anything while the IP Phone user can
hear perfectly from the PSTN user.
The 2 SER servers are setup based on onsip.org with mediaproxy configuration.
Everything works fine, except for this 1 way audio problem. Could some please
point me to the right direction to where to solve this?!?!
By the way, if my IP phone were to use a REAL IP on SER 2 and try to land calls
via SER 1, everything works perfectly!!! There will be 2 way audio. So I
would imagine this has something to do with mediaproxy or NAT issues. PLEASE
HELP!! :>
I just started installing ser the other day and the installation was smooth
and painless. I am know not quite sure about about the configuration files
and how to set them. Here is my current setup -- I have a fedora 3 linux box
obtaining a dhcp assigned address with currently no hostname set and no
FQDN-- localhost.localdomain. I signed up the other day to iptel.org to
register for a username and password -- ie, _example(a)iptel.org_
(mailto:example@iptel.org) .
So i have 2 issues. 1. What should i specify in my /etc/profile for
SIP_DOMAIN -- should i use my _example(a)iptel.org_ (mailto:example@iptel.org) or my
internal lan address or my wan address available to the outside?
2. Same question when it comes to configuring the
/usr/local/etc/ser/ser.cfg
what should be listed on the listen parameters
listen=127.0.0.1
listen=192.168.x.x or should it be the wan address
also can i use the alias
alias=_example(a)iptel.org_ (mailto:example@iptel.org) or should this be my
machines hostname.
Regards,
Alfred