Hi,
Is it possible to allow_trusted some PSTN gateways which have a dynamic
IP address?
>From what it looks like, the trusted table only allows the usage of a
source IP address. A hostname that is DNS resolvable or an IP/mask pair
would solve this.
What can be done?
Thanks,
Cristian
Hi all,
Is there support for dynamic avp/hdr names, like
$hdr($avp(s:name))
I'm getting <null> when I try this... and I've checked that
$avp(s:name) has a valid header name (Contact).
Thanks,
JF
Hi,
I realized some one is able to make call and registered to my sip
proxy while he/she is not in the subscriber table.
I couldn't find his/her username in the subscriber table, but i was
able to see him/her in the location table. I am able to see he/she
made 10 calls from my sip proxy.
Anyone have an idea on what i have done wrong? I have included the
authentication part of code here.
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest
authentication
if (!www_authorize("x.x.x.x", "subscriber")) {
www_challenge("x.x.x.x", "0");
exit;
};
consume_credentials();
save("location");
exit;
};
if (method=="INVITE") {
if (!proxy_authorize("","subscriber")) {
proxy_challenge("x.x.x.x","0");
exit;
}
consume_credentials();
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
--
Howard Tang
Hi,
I have configured apache and SER as described in presence hand book. I am
facing some problem in group resource list subscription and notification.
SUBSCRIBE sip:venkatfriends@10.187.83.218 SIP/2.0
Supported: eventlist
Subject: Subscription Request
Via: SIP/2.0/UDP 10.187.83.222:5061;rport;branch=z9hG4bK08652
Max-Forwards: 70
To: <sip:venkatfriends@10.187.83.218>
From: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489
Call-ID: 382598915012(a)10.187.83.222
CSeq: 1 SUBSCRIBE
Contact: <sip:venkat@10.187.83.222:5061>
Expires: 7200
User-Agent: XXXXXXXXXXXXXXX
Event: presence
Content-Length: 0
Accept: application/pidf+xml,multipart/related,application/rlmi+xml
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.187.83.222:5061;rport=5061;branch=z9hG4bK08652
To: <sip:venkatfriends@10.187.83.218
>;tag=c20c251a135762f9e7a5a4d8e6d1fe64-93f1
From: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489
Call-ID: 382598915012(a)10.187.83.222
CSeq: 1 SUBSCRIBE
Require: eventlist
Contact: <sip:10.187.83.218:5060>
Expires: 6200
Server: Sip EXpress router (0.10.99-dev64 (i386/linux))
Content-Length: 0
NOTIFY sip:venkat@10.187.83.222:5061 SIP/2.0
Via: SIP/2.0/UDP 10.187.83.218;branch=z9hG4bK7fa8.3ee87012.0
To: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489
From: <sip:venkatfriends@10.187.83.218
>;tag=c20c251a135762f9e7a5a4d8e6d1fe64-93f1
CSeq: 10 NOTIFY
Call-ID: 382598915012(a)10.187.83.222
Content-Length: 340
User-Agent: Sip EXpress router(0.10.99-dev64 (i386/linux))
Subscription-State: active;expires=6200
Contact: <sip:10.187.83.218:5060>
Max-Forwards: 70
Event: presence
Require: eventlist
Content-Type:
multipart/related;type="application/rlmi+xml";start="qwW388551513pPdxX1615182410";boundary="RewXdpxR742478745xA757064664";
--RewXdpxR742478745xA757064664
Content-Transfer-Encoding: binary
Content-ID: qwW388551513pPdxX1615182410
Content-Type: application/rlmi+xml;charset="UTF-8"
<?xml version="1.0"?>
<list xmlns="urn:ietf:params:xml:ns:rlmi" uri="
sip:venkatfriends@10.187.83.218" version="0" fullState="true">
</list>
--RewXdpxR742478745xA757064664--
I have created all entries in resource-list.xml and service entry in index
file.
But, the server sends NOTIFY with no resource list entries and doesn't give
any error messages. I am using full-no-failover.cfg configuration file.
Whats wrong here ?????? Can anyone help me....
-Venkat
Received the following from Klaus a few weeks ago:
Does anybody have an example of how to use sipsak with openser. Does
openser call sipsak, or is it the other way around?
Xantek, Inc. wrote:
I am sure this must have been covered somewhere, but I cannot find it.
I am trying to get openser to register with username and password
and forward a call to that server. This is for outbound sip termination
to the PSTN. openser is a proxy thus basically openser can'T do that.
Now the good news. For registration at the termination provider you can
use sipsak. For sending authentication credentials to the termination
provider openser has the "uac" module. But be aware that this module
does not always works as it does not increase the cseq when sending the
INVITE with credentials (search the archive for details). E.g. the uac
module wont work if the termination providers uses Asterisk (as
Asterisk checks the cseq)
regards
klaus
--
702-874-3833
1-866-553-3833
I have my user authentication in a MYSQL database and use openser
administrator to manage users. With every user I can add the Caller ID
which ends up in the database (I've confirmed the existence in the DB).
I'd like to add a Remote Party ID header into my calls directed to my
PSTN termination provider (this is the header they want). I have had a
look at the Auth module, and if I understood correctly, needed to add
the lines
modparam("auth", "rpid_avp", "i:13")
to obtain the rpid field from the database
and
append_rpid_hf();
to actually add the RPID field to my outgoing INVITES, etc.
however, I still can't see the header being added to the calls in
outgoing SIP traces, and I have the feeling that I'm missing something
fundamental..
I've tried also adding this line to load the credentials from the database:
modparam("auth_db", "load_credentials", "rpid")
Do I need to set the load credentials to be i:13=rpid to load the rpid
field into the avp???
It's not 100% clear from the documentation where and how these avps are
used (automatically if they're filled, or if they need to be somehow
referenced in the function that uses them) and exactly which function is
necessary (do I need the auth_db function or not?)..
Hi
I'd like to use the 'permissions' module but there is a something I need to change: the table openser.trusted has a column that matches the 'From URI' with it (from_pattern). Is there a way to change it to RURI? I mean, I don't want to analyze the header From: I want the function 'allow_trusted()' matching the RURI instead.
Is it possible?
Thanks in advance.
Bruno Machado
__________________________________________________
Fale com seus amigos de graça com o novo Yahoo! Messenger
http://br.messenger.yahoo.com/
Hi Steven,
yes, in the official distribution there is an older version of
xmlrpc-c3. Not sure why this, but we decided to build the module against
the version you currently find in all distro to avoid forcing the users
to compile/install by hand additional software.
regards,
bogdan
Steven Halerman wrote:
>Thank you for the quick response. I didn't see
>binaries for Fedora, so i downloaded the latest -
>xmlrpc-c-1.06.09 (earlier versions would not cleanly
>compile with GCC 3.4.4) from
>http://sourceforge.net/projects/xmlrpc-c but it does
>not have the xmlrpc_abyss.h file. Upon further
>digging, i noticed in server_abyss.h, a note that
>mentions
>
>Abyss contents handlers
>xmlrpc_server_abyss_rpc2_handler() and
>xmlrpc_server_abyss_default_handler() were available
>in older Xmlrpc-c, but starting with Release 1.01,
>they are not. Instead, call
>xmlrpc_server_abyss_set_handlers() to install them.
>
>Steven
>
>--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
>wrote:
>
>
>
>>Hi Steven,
>>
>>I have libxmlrpc-c3-dev version 0.9.10-4 on a
>>unstable debian and
>>compiles with no problem. The "xmlrpc_abyss.h" file
>>belong to the
>>libxmlrpc-c3-dev package:
>>/usr/include/xmlrpc_abyss.h could you check
>>if you have this hdr file installed on your machine?
>>
>>regards,
>>bogdan
>>
>>Steven Halerman wrote:
>>
>>
>>
>>>I have been trying to make mi_xmlrpc (from cvs
>>>
>>>
>>head).
>>
>>
>>>I am running Fedora Core 3 - gcc version 3.4.4
>>>
>>>I get the following error:
>>>
>>>xr_server.c:37:26: xmlrpc_abyss.h: No such file or
>>>directory
>>>mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or
>>>directory
>>>In file included from
>>>/usr/local/include/xmlrpc-c/base.h:9,
>>> from
>>>
>>>
>>/usr/local/include/xmlrpc.h:1,
>>
>>
>>> from mi_xmlrpc.h:35,
>>> from mi_xmlrpc.c:38:
>>>/usr/local/include/xmlrpc-c/util.h:20:56:
>>>xmlrpc-c/c_util.h: No such file or dir
>>>ectory
>>>In file included from
>>>/usr/local/include/xmlrpc-c/base.h:9,
>>> from
>>>
>>>
>>/usr/local/include/xmlrpc.h:1,
>>
>>
>>> from mi_xmlrpc.h:35,
>>> from mi_xmlrpc.c:38:
>>>/usr/local/include/xmlrpc-c/util.h:150: error:
>>>
>>>
>>syntax
>>
>>
>>>error before "GNU_PRINTF_A
>>>TTR"
>>>/usr/local/include/xmlrpc-c/util.h:158: error:
>>>
>>>
>>syntax
>>
>>
>>>error before "GNU_PRINTF_A
>>>TTR"
>>>mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or
>>>directory
>>>mi_xmlrpc.c: In function `mod_child_init':
>>>mi_xmlrpc.c:156: warning: implicit declaration of
>>>function `xmlrpc_server_abyss_
>>>init_registry'
>>>mi_xmlrpc.c:166: error:
>>>`xmlrpc_server_abyss_rpc2_handler' undeclared
>>>
>>>
>>(first
>>
>>
>>>use
>>>in this function)
>>>mi_xmlrpc.c:166: error: (Each undeclared identifier
>>>
>>>
>>is
>>
>>
>>>reported only once
>>>mi_xmlrpc.c:166: error: for each function it
>>>
>>>
>>appears
>>
>>
>>>in.)
>>>mi_xmlrpc.c:172: error:
>>>`xmlrpc_server_abyss_default_handler' undeclared
>>>(first
>>>use in this function)
>>>make[1]: *** [mi_xmlrpc.o] Error 1
>>>
>>>what versions of
>>> libxml
>>> libxmlrpc-c3
>>> libxmlrpc-c3-dev
>>>did you use to successfully compile?
>>>
>>>Thanks in advance
>>>Steven
>>>
>>>
Hello,
I have a problem: When someone begins a call from PSTN to my voip
network, the session interrupts after 30 secconds. The UA don't receive
the ACK message(6)[the ACK doesn't arrive].The scheme below indicate this
situation.
P.S. This situation happens just with one type of equipment: atcom
AG468; with other equipment (atcom AT320ED, audiocodes, atcom AG168)
all works fine.
Very strange.
Thunks.
|cisco-gateway| | openser | |UA's|
X.X.X.X - public ip adress eth0-X.X.X.Y - public IP 172.17.6.0/24
| eth1-172.17.6.2 |
| Invite | Invite |
1 |------------------------------->|----------------------------->|
| Trying | Trying |
2 |<-------------------------------|<-----------------------------|
| Ringing | Ringing |
3 |<-------------------------------|<-----------------------------|
| PRACK | |
4 |------------------------------->| |
| 200 ok | 200 ok |
5 |<-------------------------------|<-----------------------------|
| ACK | |
6 |------------------------------->| |
|................................|..............................|30 s
| Bye | Bye |
7 |<-------------------------------|<-----------------------------|
| 200 OK | |
|------------------------------->| |
| | |
--
Best regards,
Ion Minzu,
Specialist Tehnologii Informationale,
Administrator de sistem al Centrului de certificare,
Administrator VoIP,
I.S."Centrul de Telecomunicatii Speciale",
tel:250-517 (office), 069501208 (mob), 382869185 (ICQ)
mailto:ion.minzu@cts.md
FYI
---------- Forwarded message ----------
From: Klaus Fleischmann <kgfleischmann(a)t-online.de>
Date: Jan 16, 2007 7:01 PM
Subject: [kphone-users] New KPhone in the CVS
To: kphone-devel <kphone-devel(a)lists.sourceforge.net>, kphone-users
<kphone-users(a)lists.sourceforge.net
>
Hi,
last night I uploaded new code for kphoneSI, the most important features
are:
- SIMPLE presence: presence server support
(PUBLISH, subscription to watcherlist)
- SIMPLE presence end to end: After answered subsription, the buddy will
be shown in the contact window (in red) even if he not subscribe
us too.
- new Call Widget design
- help file and dtmf panels with own window
- the last error message will be shown in the call widget instead of
alarm windows
- some more ALSA improvements
- document and man page update
If you are not used to CVS-downloads, see
http://sourceforge.net/docman/?group_id=47742
or
http://sourceforge.net/docman/display_doc.php?docid=34779&group_id=47742
for details.
Yours
Klaus
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