Hi,
In the textops module doc
(http://openser.org/docs/modules/1.0.x/textops.html#AEN124) there is an
example that demonstrates how to replace URI in the header "To" with the
message URI:
# replace the uri in to: with the message uri (just an example) if (
subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {};
After I've fixed a small mistake (instead of /t: I put /To:) it worked.
But I don't understand how does it work. It looks like "u" is
substituted by URI from R-URI but I didn't find such core variable.
Could somebody explain me?
Thank you in advance,
Leonid Fainshtein
Dear SER Users,
I've a problem with Serweb and I hope somebody could help me out
I know this problem was already talked about but i never saw a solution to
it, so I'll post it again hoping somebody will know how to solve this
mistake.
So the problem is that when i try to put serweb working in my firefox
browser the following error is shown:
Forbidden
You don't have permission to access /serweb/index.html on this server.
Additionally, a 403 Forbidden error was encountered while trying to use an
ErrorDocument to handle the request.
------------------------------
Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80
I think i installed everything correctly.
I'll show some of the configurations I've made
#1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M because of
the Serweb extension classkit.so
…
memory_limit = 12M
#2 - I used pear to install classkit.so
pear install <path to classkit-0.4.tgz>
#3 - In "php.ini"
…
;;;;;;;;;;;;;;;;;;;;;;
; Dynamic Extensions ;
;;;;;;;;;;;;;;;;;;;;;;
…
extension=classkit.so
#4 - In "/serweb-0.4/config/config_data_layer.php":
…
$config->data_sql->host[$i]['host']="<IP MySQL Server>";
…
$config->data_sql->host[$i]['user']="<DB User>";
$config->data_sql->host[$i]['pass']="<User password>";
..
#5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php"
$config->root_path="/root/instalacao/serweb- 0.9.4/html/";
#6 - I created a log file
touch /var/log/serweb
chown apache /var/log/serweb
#7 - I created a folder for SIP domain
mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of SER)
#8 - I copied the files and folder from the default folder
("/root/installation/serweb-0.4/html/domains/_default/") to the one i
created.
#9 - I specified the SIP domain in the file
"/root/installation/serweb-0.4/config/set_domain.php":
…
$config->domain = "xxx.xxx.xxx.xxx(IP of SER)";
…
#10 - I enabled logging in file "/root/installation/serweb-0.4
/config/config.php":
…
$config->enable_loging = true;
$config->log_file = "/var/log/serweb;
#11 - In /etc/httpd/conf/httpd.conf I added an alias
…
<Directory "/var/www/html">
Options Indexes FollowSymLinks
AllowOverride None
Order deny,allow
Allow from all
</Directory>
<IfModule mod_userdir.c>
UserDir disable root
#UserDir public_html
</IfModule>
Alias /serweb/ "/root/instalacao/serweb-0.9.4/html"
<Directory "/root/instalacao/serweb-0.9.4/html">
Options Indexes MultiViews
Options +FollowSymlinks
AllowOverride None
Order Deny,Allow
Allow from all
</Directory>
...
#12 - I then put some commands for Apache to begin when computer would start
chkconfig httpd on
service httpd start
#13 - I introduced the following URL
http://<IP of SER>/<alias i wrote in httpd.conf>/
http://xxx.xxx.xxx.xxx/serweb/
And then that HTTP 403 error shows up
I really hope somebody can help me because I'm doing a VoIP project for my
school and I'd really like to have everything done in time.
Thank you for your attention,
Roberto Lopes
Hi all,
I am trying to get a better understanding of the call flow -
specifically for distant users. In our current configuration we have
all of the UAs registering to our OpenSER box which manages the
authentication, and passes the call forward if allowed. Like so:
UA -> SER -> PSTN GW -> PSTN
I have been experimenting with using a mediaproxy/rtp proxy to improve
the quality of service to users beyond the network edge - effectively:
UA -> Proxy -> SER -> PSTN GW -> PSTN
While this works great, we would like to place some PSTN Gateways
closer to the edge of the network (for more than one obvious reason).
In this case we would still have a central SER instance for
authentication and accounting, but I would like for the edge proxies
to be able to send the traffic to the closest applicable PSTN GW.
How does one go about deploying something like this? What does the
call flow look like? What software components do I need in what
places? What is the glue that holds all of this together?
Thanks,
Max
--
Max Clark
http://www.clarksys.com
Dear all,
Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much.
Thank you very much to Andrey Kouprianov for replying my message. Thanks
I have some questions about the openser admin. I do anybody can help me to understand what the openser admin use for and how to install it.
These are my question:
1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser?
2. How to install the openser administrator in openser server?
I use openser1.1.0-tls and what packets does it need?
3.Is the openser administrator just used by the administrator to manage his/her server? or The openser administrator can be used like a serweb?(the client also can use it) because as I read from the reply message in this forum, the serweb can not use for openser1.1.0-tls
4. If it is not like serweb, can anybody tell me what is the suitable web application for openser ? so the client can register to in communication session by using a web application like serweb
That is all my questions. I do hope anybody can help me. Please....
Thank you very much
Regards,
Ferianto
---------------------------------
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.
Thanks, I've looked into but it works only with 1.1.x version...!!!
What's the different with avp_db_store command, may I use it to store something in my SQL DB? I'll try to do but I have some problems, he doesn't register...
I have another question: I need to work over sdp messages, is possible to do a parsing, always using the scripts language...?
Any thought?
Thanks
Davide
-----Messaggio originale-----
Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
Inviato: mercoledì 5 aprile 2006 19.15
A: D'Addelfio Davide
Cc: Jayesh Nambiar; Bogdan-Andrei Iancu; openser
Oggetto: Re: R: [Users] using AVPs to write into database
Hello,
On 04/05/06 18:01, D'Addelfio Davide wrote:
>
> Hi guys, I have a similar problem, maybe easier. I'm new in openser so
> I'd like to have some help.
>
> I'd like openser create a table in MySQL db when receive an INVITE, a
> row with from uri, method and something else, I've look into modules
> docs but I've not understanding at all...could you help me, please?
>
if you are using openser development version, then take a look at
avp_db_query() in avpops module
http://openser.org/docs/modules/1.1.x/avpops.html#AEN239
Cheers,
Daniel
> Thanks for reply
>
> Davide
>
> ------------------------------------------------------------------------
>
> *Da:* users-bounces(a)openser.org [mailto:users-bounces@openser.org]
> *Per conto di *Jayesh Nambiar
> *Inviato:* mercoledì 5 aprile 2006 15.53
> *A:* Bogdan-Andrei Iancu
> *Cc:* openser
> *Oggetto:* Re: [Users] using AVPs to write into database
>
> Hi Bogdan,
>
> Thanks for the reply and sorry for sending the earlier mail 4 times as
> yahoo was not responding peoperly.
>
> When I do an avp_write(), why does the avp name get stored in the uuid
> column?
>
> I am not using any uuid as of now. I also tried to add the username of
> the person who is calling but it again creates two rows, one with the
> value as
>
> username and other row with the value as forwarded number.
>
> I thought if there is a way to insert the username also, it wud be
> easier to check, if the incoming call for that user is to be forwarded
> or not.
>
> I get the following in my usr_preferences table after I dial 86
> followed by any number.
>
> +-----------+----------+--------+-----------+------+-----------------------------+---------------------+
> | uuid | username | domain | attribute | type | value | modified |
> +-----------+----------+--------+-----------+------+-----------------------------+---------------------+
> | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 |
> 2006-04-05 19:10:49 |
> +-----------+----------+--------+-----------+------+-----------------------------+---------------------+
>
> The script snippet is as follows:
>
> if(uri=~"^sip:86[0-9]*@") {
>
>
> strip(2);
> avp_write("$ruri", "s:callfwd");
> #avp_write("$from/username", "i:999");
> #avp_db_store("i:999", "i:/usr_preferences");
> avp_db_store("s:callfwd", "s:callfwd/usr_preferences");
> sl_send_reply("200", "OK");
> exit;
> };
>
>
> */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* wrote:
>
> Hi,
>
> you need two steps:
> 1) write the ruri (after strip) into an AVP - use avp_write()
> 2) write the avp into the db - use avp_db_store()
>
> regards,
> bogdan
>
> Jayesh Nambiar wrote:
>
> > Hi all,
> > I am trying to implement the call forwarding feature in openser. The
> > forwarding part is fine. But I have the follwing scenario:
> > Whenever a user needs to change his forwarding number, he should be
> > able to do so from his end device only.
> > For eg: He first presses 86 and then the 10 digit number to be
> > forwarded. Here the 86 has to be stripped and the 10 digit number
> > should be inserted into the call forward value column of the
> > preferences table.
> > Also if he wants to remove the forwarded number, he can do so by
> > pressing say for eg: 87. on receiving this number openser should
> > delete that value from the
> > table.
> > Is this possible. I tried to do the following, but somehow it
> does not
> > change the
> > value.
> > if(uri=~"^sip:86[0-9]*@") {
> > if(avp_db_load("$from/username", "s:callfwd")) { #check
> > if call-fwd feature is enabled for the user
> > log(1,"AVP condition returned true");
> > strip(2);
> > avp_write("$ruri", "s:callfwd");
> > avp_print();
> > log(1,"AVP written");
> > sl_send_reply("200", "OK");
> > exit;
> > };
> > };
> > Is avp_write the proper method or I guess avp_db_store can also help
> > me. The avp_print() function also does not show me anything in the
> > log. Are there any
> > logical mistakes or I have mis-interpreted the syntax of avpops
> functions.
> > Please help me in thsi regard.
> > Thanks a lot in advance.
> > Jayesh.
> >
> > ------------------------------------------------------------------------
> > Jiyo cricket on Yahoo! India cricket
> >
> > Yahoo! Messenger Mobile
> >
> > Stay in touch with your buddies all the time.
> >
> >------------------------------------------------------------------------
> >
> >_______________________________________________
> >Users mailing list
> >Users(a)openser.org
> >http://openser.org/cgi-bin/mailman/listinfo/users
> >
> >
>
> ------------------------------------------------------------------------
>
> Jiyo cricket on Yahoo! India cricket
> <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…>
> Yahoo! Messenger Mobile
> <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…>
> Stay in touch with your buddies all the time.
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
Hi
i am new to ser,
iam going through the list,
i found some of the similar queries
http://lists.iptel.org/pipermail/serusers/2003-October/003132.html
I have used the config file
changed according to my setup
and iam able to pick up messages from Asterisk also
But i have some problem here is
1. i have registered user 8888 in my SER, when i dial 8888
its going to voice mail, how can i set the caller==calee Busy
even i have added this
if (!uri==myself) {
t_relay();
break;
};
But still no use.
2. when i dial to PSTN number, when the PSTN also go to voice mail after
certain rings
I suppose to get the voice Menu of other Side Number, but the voice go Blank
( instead of going to VoiceMenu)
when checked in my Log, i get this error in my syslog
Aug 28 18:33:55 router ser[757]: no location or no user
Aug 28 18:34:01 router ser[751]: 10 digit exp match w/leading 1
Aug 28 18:34:01 router ser[751]: route[2]:SIP-to-PSTN call routed
Aug 28 18:34:10 router ser[757]: no location or no user
Aug 28 18:34:25 router ser[758]: no location or no user
Aug 28 18:34:28 router ser[791]: ACC: call missed: from=88888 <
sip:88888@mydomain.com>;tag=1674409670, i-uri= sip:1xxxxx80369@mydomain.com,
method=INVITE, o-uri=sip:1xxxxx80369@myprovider:5060, code=408 Request
Timeout
Aug 28 18:34:28 router ser[791]: failure_route[5]
Aug 28 18:34:28 router ser[791]: route[5]: re-relay to PSTN
Aug 28 18:34:28 router ser[751]: ERROR: t_should_relay: status rewrite by
UAS: stored: 408, received: 487
Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1
Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed
Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1
Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed
Aug 28 18:34:40 router ser[751]: no location or no user
any suggestions
Ram
I'm sceptic about the use of functions that replace parts of SIP messages,
because that way, will Ser still keep record of transactions? Wont those
replacements affect Ser to recognise to whom route subsequent messages?
Regards,
Ricardo.
Alexandr Dubovikov wrote:
> On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote:
>> I need to rewrite the From part of SIP INVITE messages before sending
>> them to my IP Telco, or else they wont accept my calls!
>> Anyone knows how can I do that?
>
> replace or subst
>
> the sample you can find in README file of textops module.
>
>
>> Thanks,
>>
>> Ricardo.
>> _______________________________________________
>> Serusers mailing list
>> Serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>
