I haven't been involved with OpenSER for a while.
Can someone tell me if OpenSER now supports DNS SRV lookups when it routes
calls? Which mnimum version supports this?
Thanks.
Doug.
Dear all
I changed the used Software for my Presence Agent after couple of time
and have problems to use presence with XCAP authorization.
The debug messages
PRESENCE_XML:get_rule_node:ERROR didn't found identity tag
PRESENCE_XML:get_rule_node:ERROR didn't found identity tag
PRESENCE: build_str_hdr: ERROR bad status flag= 136681324
PRESENCE:send_notify_request:ERROR while building headers
PRESENCE:notify: ERROR send Notify not successful
PRESENCE:update_subscribtion: Could not send notify for presence
appear.
I have no idea what I did wrong. I am still using my XCAP-Lite server
version 1.0.0 and adjusted the config to using with the presence_xml
module, not only presence. Furthermore I did a complete database update,
so all tables are in the current version.
One client requests authorization which is granted by the other. The
accordant XCAP data is right in the xcap_xml table. But still, no NOTIFY
is sent :(
Does anyone experience similar problems? Where can be the issue for the
posted error messages?
Thanks as always for your help!
Best regards
Sebastian
Sebastian Schumann
Diploma Student Architecture and Design Department
Slovak Telekom, a. s.
T-Com, Operations Unit
Production and Service Division
Innovations and Enterprise Solution Subunit
Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia
Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia
+421 2 588 13332 (tel)
+421 910 643010 (mobile)
+49 175 1925928 (mobile)
sebastian.schumann(a)t-com.sk
http://www.t-com.sk <http://www.t-com.sk/>
Hello,
I am newbie to this field. We are trying to use a separate card for
handling the rtp voice traffic but use the RTPProxy (IPtel) software to
handle the NAT issues. We have been fairly successful in doing that.
The problem now is to handle asymmetric NATs. I hear one way audio.We
are not sure how to handle this when there is one machine running
rtpproxy software and another card with a different IP handling the SDP
(In nathelper.c of Openser new_ip= card IP). In the symmetric NAT
scenario this works fine. I would appreciate and pointers to modify
rtpproxy for this configuration.
We use rtpproxy 0.3, Opneser 1.1.0.
-the card only handles RTP data and has its
own GbE port
- made changes to nathelper.c nathelper.cfg
source code to account for separate IP addrs
- made changes to RTPProxy to open channels for RTPtraffic flow on the card.
-Openser,RTPProxy and the card , all 3 different Public IPs
Thanks
Nithin
Hi all.
MySql password for root:
Domain (realm) for the default user 'admin': 10.200.19.34
creating database openser ...
ERROR 1071 (42000) at line 131: Specified key was too long; max key length
is 1000 bytes
Install SERWEB tables ?(y/n):
Is there anyway to solve this issue...?
--
Best Regards,
Arshad Mehrab Memon
Hi @ all,
I wanted to use the LCR module with OpenSER 1.2.
My tables look like this:
Lcr:
PREFIX | FROM_URI | GRP_ID |
PRIORITY |
------------------------------------------------------------------------
-------------------------------------------------------------
| | 1 |
1 |
Gw:
GW_NAME | GRP_ID | IP_ADDR | PORT |
URI_SCHEME | TRANSPORT | STRIP | PREFIX
------------------------------------------------------------------------
------------------------------------------------------------------------
------------------------------------------------------------------------
-------------
Gw 1 xxx.xx.xxx.xx
5060
Gw_grp:
GRP_ID GRP_NAME
----------
----------------------------------------------------------------
1 VIH
I always get the following error:
CALL: Call to international number <sip:0004312345586@server.com> from
<sip:1004@server.com>
0(23072) LCR: no gateways found m=INVITE
ruri=sip:0004312345586@server.com f=sip:1004@server.com
t=sip:0004312345586@server.com ip=xxx.xx.xxx.xxx
Does anybody can help me?
Thanks
Martin
Hi
I'm trying to do some call forking but I can't manage to make it work well.
I explain quickly what it is :
If user A call user B and user B is registered more than once (on different
system), the proxy will fork natively to all registered user B.
What I want to do now is if user A call user B then the proxy fork to user
C, user D and so on.
I have found the command append_branch("sip:userC@domain").
The thing is that it is not working well.
For the first time (when the server restart), I make a call from user A,
only user B receive the call, then ignored the call from the user B and
wait for about 15sec then the call ring on user C and D.
I make the 2nd try, user A call B, B, C & D receive the ring.
Then I wait some minute, the problem came back again
what is wrong with the first call, a bug or configuration error ?
please help
Bernard
here is my config file :
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
children=4
listen=192.168.17.107
port=5060
# ------------------ module loading ----------------------------------
#set module path
mpath="/usr/local/lib/openser/modules/"
loadmodule "mysql.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
# ----------------- setting module-specific parameters ---------------
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("usrloc", "db_mode", 3)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
if (!mf_process_maxfwd_header("10"))
{
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 )
{
sl_send_reply("513", "Message too big");
exit;
};
if (!method=="REGISTER")
record_route();
if (loose_route())
{
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself)
{
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
};
if (uri==myself)
{
if (method=="REGISTER")
{
route(2);
};
if (method=="INVITE")
{
route(3);
};
lookup("aliases");
if (!uri==myself)
{
append_hf("P-hint: outbound alias\r\n");
route(1);
};
if (!lookup("location"))
{
sl_send_reply("404", "User not found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1]
{
if (!t_relay())
{
sl_reply_error();
};
exit;
}
route[2]
{
if (!www_authorize("etraliSIP.fr", "subscriber"))
{
www_challenge("etraliSIP.fr", "1");
exit;
};
save("location");
exit;
}
route[3]
{
if (!proxy_authorize("etraliSIP.fr", "subscriber"))
{
proxy_challenge("etraliSIP.fr", "1");
exit;
};
if (from_uri=="sip:leb@etraliSIP" && to_uri=="sip:patrice@etraliSIP"
&& !search("P-hint: ok call forking"))
{
append_hf("P-hint: ok call forking\r\n");
append_branch("sip:test@etraliSIP");
append_branch("sip:tony@etraliSIP");
};
}
*********************************
This message and any attachments (the "message") are confidential and intended solely for the addressees.
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Messages are susceptible to alteration.
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Hi all,
I received following error while configuring openser with MYSQL.
MySql password for root:
Domain (realm) for the default user 'admin': 192.168.1.10
creating database openser ...
ERROR 1007 (HY000) at line 1: Can't create database 'openser'; database
exists
Install SERWEB tables ?(y/n):n
Any help would be highly appreciated.
--
Best Regards,
Arshad Mehrab Memon
On 08/20/07 21:05, Kelvin Williams wrote:
> THANK YOU!!!!!
>
> The only problem here is the vendor, Arris Interactive believes their
> products are the greatest. And "officially do not support SIP Deployment."
>
> Could you point me in the direction of some documentation as to how I can
> drop the 183 and generate a 182 instead?
>
generating other reply code is a bit hard, needs some coding. Dropping
183 is easy:
http://openser.org/dokuwiki/doku.php/core-cookbook:devel#drop
Cheers,
Daniel
> Thanks,
> kw
>
> -----Original Message-----
> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
> Sent: Monday, August 20, 2007 12:30 PM
> To: kelvin-lists(a)williamschadwell.com
> Cc: users(a)openser.org
> Subject: Re: [OpenSER-Users] Changing Audio Ports
>
> Hello,
>
> the sip device is broken, should update the destination port and IP from
> the 200 ok reply. The best is to ask the vendor to fix it. Think about
> para;lel forking, there could be many 183 coming from different
> addresses and ports, what would do the device?
>
> Temporary solution is to drop any 183 or to use perl substitutions for
> such replies (all to be done in onreply_route).
>
> Cheers,
> Daniel
>
> On 08/18/07 07:24, kelvin-lists(a)williamschadwell.com wrote:
>
>> In a previous query I asked if someone could shed some light on as to why
>> my endpoints do not receive any audio when the call is redirected to an
>> announcement server.
>>
>> After a lot of testing, I believe I have found the problem.
>>
>> When I initiate a call from my end point the end point advises the callee
>> as to the port the RTP traffic will be present. When the call is handed
>> to my PSTN gateway the Gateway responds with its port for RTP in a 183
>> Session Progress.
>>
>> When that call fails (due to timeout) we want to send it over to Asterisk
>> where an announcement will be played to the caller--however the caller
>> never hears it. The traces show that Asterisk advertises its RTP on a
>> different port that that of the PSTN Gateway. Some of my endpoints (Cisco
>> IP Phone 7940 and Sipura devices) see and listen for the audio on the new
>> advertised port, however my Arris EMTAs do not, it appears as though they
>> are still "tuned in" to the original audio port advised by the PSTN
>> gateway.
>>
>> My question, is is possible to strip away the "m=audio 22040 RTP/AVP 0 8
>> 18 101." from the SIP message? I would like to strip it away in the event
>> of a 183 from my gateway (that advertises the port), but pass it when the
>> call is actually answered.
>>
>> If it is not possible to strip the RTP port information away from the
>> message, what would be the best way in handling a situation like this.
>>
>> Many thanks in advance.
>> kw
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
>
>
*********************************
This message and any attachments (the "message") are confidential and intended solely for the addressees.
Any unauthorised use or dissemination is prohibited.
Messages are susceptible to alteration.
France Telecom Group shall not be liable for the message if altered, changed or falsified.
If you are not the intended addressee of this message, please cancel it immediately and inform the sender.
********************************