Hello,
I'm using a Sipura device to register to my ser server. When I use a
Softphone, the contact-information in the location table is OK, but when I
use my Sipura device, ser saves the contact information with the internal IP
address of the sipura. Obviously ser doesn't find it when I want to make a
call to a Sipura-Phone.
Any idea what might be the problem?
THanks!
Sebastian
Dear List,
I can register my ATA'S (Some are NATted and some ara Port Forwarded) to SER (on Public IP) and i use MySQL auth/acc together with Mediaproxy. All users can talk to others, and call them , receive calls from others no problem. But most of the time, they seem unregistered and this happens suddenly, i'm sure that Internet Connection is fine with SER Server and also clients are connected through Cable Modem or ADSL Connection.
I unplug or unpower some ATA's, wait a bit like 30 minutes, the devices becomes cold or what's happening else, they can register and works perfect... But when i connect them for hours, they unregister themselves, i do not know what happens... I've tried both Zyxel P2002's with several firmwares, Zyxel P2000Ws, Cisco ATA186's and some LinkSYS Pap2-NA's but all same... Is there any chance to be something wrong in SER settings or Operating System ?
Thanks,
Ozan
harry gaillac wrote:
>hello,
>
>use database apache serweb logs to find problem
>--- Chris Mason <lists(a)masonc.com> a écrit :
>
>
Harry:
I found the problem as shown below. The applicication is reuqesting a
column that does not exist in the database. I checked the contents of
scripts/ser_mysql.sh from ser-0.9.3_src.tar.gz
<ftp://ftp.berlios.de/pub/ser/0.9.3/src/ser-0.9.3_src.tar.gz> and find
that column does not exist in the subscriber table. There's no way I can
get it to work. Is the ser distribution broken or the serweb?
Chris
Query as executed by user ser
#######################################################################
mysql> select phplib_id from subscriber where username='admin' and
password='heslo' and perms='admin' and domain='mason.home'
-> ;
ERROR 1054 (42S22): Unknown column 'perms' in 'where clause'
#######################################################################
create table statement from scripts/ser_mysql.sh
#######################################################################
#
# Table structure for table 'subscriber' -- user database
#
CREATE TABLE subscriber (
phplib_id varchar(32) NOT NULL default '',
$USERCOL varchar(64) NOT NULL default '',
domain varchar(128) NOT NULL default '',
password varchar(25) NOT NULL default '',
first_name varchar(25) NOT NULL default '',
last_name varchar(45) NOT NULL default '',
phone varchar(15) NOT NULL default '',
email_address varchar(50) NOT NULL default '',
datetime_created datetime NOT NULL default '0000-00-00 00:00:00',
datetime_modified datetime NOT NULL default '0000-00-00 00:00:00',
confirmation varchar(64) NOT NULL default '',
flag char(1) NOT NULL default 'o',
sendnotification varchar(50) NOT NULL default '',
greeting varchar(50) NOT NULL default '',
ha1 varchar(128) NOT NULL default '',
ha1b varchar(128) NOT NULL default '',
allow_find char(1) NOT NULL default '0',
timezone varchar(128) default NULL,
rpid varchar(128) default NULL,
domn int(10) default NULL,
uuid varchar(64) default NULL,
UNIQUE KEY phplib_id (phplib_id),
PRIMARY KEY ($USERCOL, domain),
KEY user_2 ($USERCOL)
) $TABLE_TYPE;
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: netconcepts_anguilla(a)yahoo.com
I've been using asterisk for voicemail, but I'm curious about peoples'
opinions about SEMS since 0.9.0. It would be nice to run something more
tightly integrated with SER, but I don't want to lose the flexibility that
asterisk offers. Opinions?
Dan
Hello:
We recently upgraded to RedHat ES 4 and SER 0.9.3. The upgrade went
well however SER runs ok for a few hours then crashes. We have not been
able to
identify the source of the problem. The obvious thing would be to
downgrade the OS
but that is also the least desirable choice. Is there a list of
packages that SER is
dependent upon? I'm suspecting a MySQL problem but I cannot confirm this.
Right now we have the following MySQL packages installed:
mysql-4.1.10a-2.RHEL4.1
MySQL-shared-compat-4.0.18-0
perl-DBD-MySQL-2.9004-3.1
php-mysql-4.3.9-3.6
mysql-server-4.1.10a-2.RHEL4.1
Thanks,Steve
--
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
Hi all. Thanks to Jan and Andres for the help.
--- ------
Things are working better now.
I did some tests and now I´m sure. Asterisk don´t give a dam to "a=direction:active". It only
cares about the NAT="yes" parameter in "sip.conf" file - or sip_buddies table for realtime users... -
For me its ok. My SER/Asterisk host has a public IP. So, it works. I could debug sip and rtp and I saw it erroring "could not find network to send RTP" and when the UA send it first packet...TA-DA!! Everything works fine! RTP media path goes like oil!
----------------------------------
I disabled the SDP fix from my "ser.cfg" so I can make my NATed UAs - behind the same NAT - talk to each other and establish the media path
based on internal values that, in this case, are true values.
To fix when a public UA calls a NATed one, I use a t_on_reply to fix the 200-0K of all UAs that are being called from public UAs. When the INVITE cames from a NATed UA, I use another route - for t_on_reply() - and it don´t fix the 200-OK and my first case - media between NATed - keeps working...
Ok. But when a NATed UA from ANOTHER NATED NETWORK - calls a NATed UA from my network I got problems. Problems because my "ser.cfg" won´t deal with SDP values and the internal IP:Port for media path will be kept. With this, I will never be conversation between then.
DO ANYBODY HAS ANY IDEA/PRODUCTION CASE TO DEAL WITH THAT?
----------------------------------------------------------
Thanks in advance.
Ricardo Poppi.
Date: Fri, 8 Jul 2005 20:51:58 +0200
From: Jan Janak <jan(a)iptel.org>
Subject: Re: [Serusers] NAT - Lots of flavours...
To: Ricardo Poppi <rpoppi77(a)giro.com.br>
Cc: serusers(a)lists.iptel.org, sobomax(a)portaone.com
Message-ID: <20050708185158.GY6497(a)localhost.localdomain>
Content-Type: text/plain; charset=iso-8859-2
Hello, comments inline.
On 04-07-2005 19:44, Ricardo Poppi wrote:
>>
>> Hi list,
>>
>> I´m trying to put to work a NATed environment and want to share some
>> information and request some I don´t realized yet.
>>
>> I use an asterisk gateway, with a public IP, working really fine for UAs
>> with public IPs. At the same machine I runs SER that receives all SIP
>> messages and handle when it should go to a SIP UA or to asterisk,
>> rewriting the port (to the one asterisk uses) and sending to it. I don´t
>> replicate register to asterisk, and use the user accounts as "peer",
>> instead of "friends".
>>
>> My ser.cfg is using the "force_rport()" and "fix_nated_contact()" for
>> every REGISTER it receives from nat UAs - I know when it comes from a
>> NATed UA using nat_uac_test("2").
>>
>> Every INVITE that comes from NATed UA passes through a
>> "fix_nated_sdp("2"), that rewrites the IP address of SDP headers. Using
>> a onreply route I fix the 200 OK INVITE message, just in case that the
>> NATed UA is on the called side.
>>
>> The UAs I´m using are X-Lite, Clipcomm CP-100 IP Phone, and Grandstream
>> HT-488.
>>
>>
>> Below I wrote the different kinds of configuration into the UA and in
>> ser.cfg, and the results I got:
>>
>>
>> 1) Using without touching the UA - It don´t know it is a NATed UA.
>> -----------------------------------------------------------------------------------------------------------------------------
>>
>> All REGISTER are treated ok because the force_rport make SER respond to
>> the register on the same external IP:Port it received. On the same hand,
>> it stores the right URI into the location database making the UA receive
>> the subsequent INVITES or other SIP messages through the external IP:Port.
>>
>> The INVITES that comes from NATed UA have their SDP IP address rewriten
>> by SER and the external IP takes place. But the port is kept the
>> internal value, so when the called UA tries to reach the
>> External_IP:Internal_port the NAT/Firewall probably block/drops the
>> packets, and the result is a one-way audio - The one-way audio is
>> probably due to the right value that comes from the SDP headers of the
>> called UA - asterisk -, that has a public IP.
>
>
I just quickly looked into nathelper sources and it looks like it can
only rewrite the port number in SDP if you run force_rtp_proxy,
fix_nated_sdp seems to change the IP address only. I CCed to Maxim,
maybe he could clarify this better than myself.
>> 2) a=direction:active
>> ----------------------------------
>>
>> If I add into ser.cfg a "fix_nated_sdp("1")" command, it will add the
>> "a=direction:active" parameter to SDP header of INVITE that comes from
>> NATed UAs. I saw that it´s happening but the asterisk seems to not
>> understand that and don´t expect for the first RTP packet to get the
>> IP:Port information of the media. A one-way audio is the result of that.
>> The asterisk is probably sending RTP packets to the
>> Ext_IP:Internal_port, and the firewall is blocking the packets.
>
>
If asterisk does not support symmetric RTP then you will have to put
the rtpproxy between the user agent and asterisk and call
force_rtp_proxy instead of fix_nated_sdp in the script. I am not sure
if I remember it correctly, but I think that asterisk should support
symmetrict rtp, so maybe the problem is in fix_nated_sdp function
which does not alter media ports.
---------
Does anyone on the list know if asterisk supports symmetric RTP ? In
other words, can asterisk interpret a=direction:active parameter from
SDP and send media to the source IP and port of the incoming media
stream, instead of the IP and port from SDP ?
---------
You can also try to put rtpproxy between user agents and asterisk and
call force_rtp_proxy instead fix_nated_sdp. It's not the best
solution, but this way you could verify if the problem is in unaltered
port number in SDP.
Jan.
Hello,
plase (CC) the list in your replies.
serweb-0.9.3 match tables of ser-0.9.3
But there are a little confusion in db schema of ser-0.9.3 and ser-cvs.
Some tables have diferent names.
serweb-cvs should also match tables of ser-cvs, but I didn't check it in
last days. Note that cvs version is not oficial release. And is not
recomended to mix develeper version of serweb from cvs and oficial
release of ser.
Karel
harry gaillac napsal(a):
> Hello Karel,
>
> why serweb-0.9.3 don't match ser tables?
>
> I think you read my posts
>
> Harry
> --- Karel Kozlik <karel(a)iptel.org> a écrit :
>
>
>>Hello,
>>there are two ways:
>>- install serweb 0.9.3 (recomended)
>>- change cofiguration of your version of serweb to
>>fit db schema of
>> ser 0.9.3. Change the names of tables in
>>config_data_layer.php
>>
>>Karel
>>
>>Timur V. Elzhov napsal(a):
>>
>>>Dear SER experts,
>>>
>>>I've downloaded and configured SER v0.9.3, as well
>>
>>as serweb, the
>>
>>>head CVS version. I succeeds by serweb with
>>
>>subscribtion, logging
>>
>>>on, and changing my own details. But the red
>>
>>message appears on my
>>
>>>user page:
>>>
>>> "DB Error: no such table".
>>>
>>>However, again, there is now problem with actions
>>
>>mentioned just above.
>>
>>>I'm not familiar with PHP, but I found that many
>>
>>php files under
>>
>>>"serweb/data_layer/" directory contain the same
>>
>>"if" statement comprising
>>
>>>the "DB_ERROR_NOSUCHTABLE" constant:
>>>
>>> if (DB::isError($res)) {
>>> if ($res->getCode()==DB_ERROR_NOSUCHTABLE)
>>
>>return true; //expected, table mayn't exist in
>>installed version
>>
>>> else {log_errors($res, $errors); return
>>
>>false;}
>>
>>> }
>>>
>>>- it may be useful ...
>>>
>>>How should I locate & resolve this problem?
>>>
>>>Much thanks.
>>>
>>>
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
>
>
>
>
>
>
> ___________________________________________________________________________
> Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
> Téléchargez cette version sur http://fr.messenger.yahoo.com
Hi there guys,
This is my first question to the list, hopefully it will be a painless issue for everyone else :)
I am trying to configure SER to have my users register to it and be able to call themselves by username (SIP/cdoyle@ser), and also call PSTN numbers and have them forwarded to an asterisk PBX for PSTN termination.
Everything seems to be setup properly to me, but when I try to make a call to the PSTN, I get "DEBUG: sl_filter_ACK : local ACK found -> dropping it!"
My config file is at http://caves.narshe.net/ser.cfg
Thanks
Chris Doyle
Hi,
I am trying to give my users the option of disabling caller ID if they so
choose by dialing a code. I can tack on rpid using append_rpid_hf() just
fine, but for some reason my privacy flags seem to be ignored. Perhaps I am
not processing them properly?
One observation:
I've noticed most people on the list using the format:
append_rpid_hf("<sip:","@localhost;
user=phone>;party=calling;screen=no;privacy=full")
When use this format my tcpdumps show:
Remote-Party-ID: <sip:sip:2125551212@my.sip-domain.com@localhost;
user=phone>;party=calling;screen=no;privacy=full")
I have to use:
append_rpid_hf("<","; user=phone>;party=calling;screen=no;privacy=full")
...to get it to look correct in the headers. This strikes me as odd that
everyone else uses the former and I must use the latter.
I am using ser-0.9.3 (Soon to upgrade to 0.9.3!!) and the gateway is a Cisco
AS5350. I've looked as Cisco DOCS and the header seems to be structured
properly. I'm sure it is something stupid, but I am scratching my head on
this one.
Any advice?
Thank you.
Dan
