Hi,
im using Linphone, when im about to connect using ipv6 address, i
found that my softphone keep sending REGISTER message;
-------------------------------------------------
11:54:49.475240 IP6 2004:fa22:337:4::82.sip > 2004:fa22:337:4::81.sip:
SIP, length: 400
`......@ ..".7.......... ..".7.................aREGISTER
sip:[2004:fa22:337:4::81] SIP/2.0
Via: SIP/2.0/UDP [2004:fa22:337:4::82]:5060;branch=z9hG4bK679317669
From: <sip:2002@[2004:fa22:337:4:]:81>;tag=618703948
To: <sip:2002@[2004:fa22:337:4:]:81>
Call-ID: 1861704303@2004:fa22:337:4::82
CSeq: 1 REGISTER
Contact: <sip:2002@[2004:fa22:337:4::82]:5060>
Max-Forwards: 70
User-Agent: Linphone-1.7.1/eXosip
Expires: 900
Content-Length: 0
11:54:49.475635 IP6 2004:fa22:337:4::81.sip > 2004:fa22:337:4::82.sip:
SIP, length: 437
`......@ ..".7.......... ..".7..................SIP/2.0 200 OK
Via: SIP/2.0/UDP
[2004:fa22:337:4::82]:5060;branch=z9hG4bK679317669;received=2004:FA22:337:4:0:0:0:82
From: <sip:2002@[2004:fa22:337:4:]:81>;tag=618703948
To: <sip:2002@[2004:fa22:337:4:]:81>;tag=9bf42f041e5ea2673c94775a0faadb8b.a53a
Call-ID: 1861704303@2004:fa22:337:4::82
CSeq: 1 REGISTER
Contact: <sip:2002@[2004:fa22:337:4::82]:5060>;expires=892
Server: OpenSER (1.3.0-notls (i386/linux))
Content-Length: 0
-------------------------------------------------------------
the softphone continue to sending REGISTER even after receiving 200 OK...
until the server send 400 Bad Request;
----------------------------------------------------------
11:58:29.577922 IP6 2004:fa22:337:4::81.sip > 2004:fa22:337:4::82.sip:
SIP, length: 486
`......@ ..".7.......... ..".7..................SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP
[2004:fa22:337:4::82]:5060;branch=z9hG4bK728798624;received=2004:FA22:337:4:0:0:0:82
From: <sip:2002@[2004:fa22:337:4:]:81>;tag=618703948
To: <sip:2002@[2004:fa22:337:4:]:81>;tag=9bf42f041e5ea2673c94775a0faadb8b.3e2a
Call-ID: 1861704303@2004:fa22:337:4::82
CSeq: 2 REGISTER
Contact: <sip:2002@[2004:fa22:337:4::82]:5060>;expires=873
P-Registrar-Error: Invalid CSeq number
Server: OpenSER (1.3.0-notls (i386/linux))
Content-Length: 0
-------------------------------------------------------
PS: the softphone works fine on ipv4
any suggestion would be very helpful...
I have a situation where i have:
Asterisk => PSTN gateway
openSER => registrar/proxy
PhoneA -> openSER -> Asterisk -> PSTN phoneB
my phones connect to openser...
calls are sent out to the PSTN using Asterisk...but when phoneA hangs
up, Asterisk does not receive the hang up signal.
If PSTN phone B hangs up, asterisk receives the hangup signal.
I'm reading forums but not receiving any real solution? is there anyway
of fixing this?
-
Gerard
I am trying to get presence working on a Snom 320 with OpenSER v
1.3.2.However the phone is not setting the sla parameter in the dialog
Event which
is required by OpenSER. However the parameter is set on this example from
the Snom Wiki
http://wiki.snom.com/SIP_Traces
Any ideas how to enable it?
TIA
Regards
Jon
Hi there,
I was using pua_bla and presence modules successfully with OpenSER 1.3.0
I've upgraded to OpenSER 1.3.2 and now BLA/Presence functionality is broken.
Now I'm getting the error:
ERROR:pua_bla:bla_handle_notify: content length= 0
Is it possible that something change regarding BLA/Presence in 1.3.2?
Nothing in the change log...
Thanks in advance.
Pablo Saro
psaro at google dot com
Hi,
I am working on performance test on Openser, and I am getting:
ERROR:tm:sip_msg_cloner: no more share memory
ERROR:tm:store_reply: failed to alloc' clone memory
The share memory allocated is 512M but the error still occurs when
concurrent call reached > 100.
Is this normal?
Is there any benchmarking number that can help to determine the amound of
shared memory needed by Openser?
Thanks,
Mark
Hi, all
We are implementing a failure route upon t_relay() to UASs.
This failure route is intended to a Caller who's destination UAS is not
responding at all, till the request gets timed out, which can be arguably
interpreted as UAS' internet is down. The purpose is to re-route incoming calls to
any alternatives; i.e. cell pnone or other landline # when the remote user site lost
netwrok connectivity.
For the scenario above, t_check_status("408") would be the best for determining?
Any other better methods or test functions in Openser?
Greatly appreciate experienced' guidance.
Thanks in advance.
John Chahn Kim
Hello, I'm trying to test Mediaproxy 2.0 however the documentation seems
vague in the following
- Can I test this on Debian Etch kernel version 2.6.18 ?
- The documentation says it cannot be used in combination with any version
of OpenSER older than 1.4. The current stable version of OpenSER available
is 1.3.2. do I have to run Mediaproxy 2.0 on an unstable version of
OpenSER ?
- Also it reads on the documentation: To display the history of the media
streams CDRTool 6.5 or higher is required but only 6.4.1 is available on the
download site.
- What is the procedure to install Mediaproxy 2.0 once you download it ?
I'm not familiar with Python and the documentation mainly talks about the
requiremnts but there is no explanation on how to install it.
I copied config.ini.sample to config.ini and edited config.ini, copied
debian/mediaproxy-dispatcher.init to /etc/init.d/mediaproxy-dispatcher. Then
I tried running media-dispatcher but I got the following error:
host.locahosts# /etc/init.d/mediaproxy-dispatcher start
Starting MediaProxy dispatcher: media-dispatcherTraceback (most recent call
last):
File "/usr/bin/media-dispatcher", line 12, in ?
from application.process import process, ProcessError
ImportError: No module named application.process
already running.
Thanks
John
hi , thank you for the answer Daniel, but I am a little confused
if (!method=="REGISTER")
record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (has_totag()) {
if (loose_route()) {
if(method=="INVITE" && (!allow_trusted())) {
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
exit;
} else if (!check_from()) {
sl_send_reply("403", "Forbidden, use From=ID");
exit;
};
### section for asterisk ####
route[4] {
rewritehostport("192.168.10.1:5070");
t_relay();
route(1);
}
route[10] {
append_hf("P-hint: inbound->inbound \r\n");
if (uri=~"^sip:[29][0-9]{7}@") {
if (is_user_in("credentials","local")) {
route(4);
exit;
} else {
sl_send_reply("403", "Not permissions local call");
exit;
};
};
if (uri=~"^sip:1[2-9][1-9]{3}@") {
if (is_user_in("credentials","int")) {
route(4);
exit;
} else {
sl_send_reply("403", "Not permissions internal call");
exit;
};
};
if (uri=~"^sip:011[0-9]*@") {
if (is_user_in("credentials","international")) {
route(4);
exit;
ettc etc ,
inside the sip.conf
[openser]
type=friend
context=netsoluciones
insecure=very
host=localhost
disallow=all
allow=alaw
allow=g729
;(
which the best form is to authenticate the invites or not? if asterisk and openser are in the same box
where my error can be ?
regards
rickygm
----- Original Message ----
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Friday, July 18, 2008 8:16:31 AM
Subject: Re: [OpenSER-Users] openser and asterisk in the same pc
Hello,
>the authentication is required by openser -- that means there is a
>mistake in your configuration file if you don't want to authenticate
>such INVITE. The phone should respond with a new INVITE having
>credentials header -- you have to set the username and password in you
>phone as well.