Without being the one who wrote the code it seems like you either (a)
wouldn't need usrloc db_mode 1 or 2 or (b) would need to store them in
separate databases. I have never been able to get usrloc db working
properly so I am in the process of ditching it (re: Keeping binding ...
for replication, error). Not only that the aliases never seem to work
properly after a restart of SER, they always go into zombie state which
I don't believe should be normal..
my 0.02
Matt
-----Original Message-----
From: AJ Grinnell [mailto:ajgrinnell@gmail.com]
Sent: Tuesday, April 05, 2005 2:08 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] More usrloc sync questions
So I can get replication working, but I constantly am getting errors
about receiving SERs usrloc not being able to insert into the database.
My guess is because the contact is already there by the originating SER.
I have a MySQL cluster that both SERs are using. My question is this,
how can I save the contact in the SER that is receiving the contact,
without SER trying to write the contact to the DB?
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hello list
can i use radiusclient-nf-0.5.0 with ser-0.8.14
thanks
Kamran
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Hey guys,
Got serweb reporting N/a for "Length of call" and "Hangup", so i've checked
a few google howto's and example configs, however still not getting any.
I get the initial INVITE and an ACK in the same second, but when the call
ends, nothing.
I have the following in my ser.cfg:
# -- acc params --
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1 )
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("rr", "enable_full_lr", 1)
# account completed transactions via syslog
setflag(1);
setflag(2);
if(method=="INVITE" || (method=="ACK") || method=="BYE" ){
setflag(1);
setflag(2);
};
Barry
Hi all.
For some reason this ACK looped and I have no idea why. It's very suspicious
that the Record-Route in the second ACK has a branch=0 tag.
NOTE: 10.3.0.221 <http://10.3.0.221> is the IP of my ser proxy. It sits
behind a Cisco 3600 so the IPs have been rewritten to
10.3.0.221<http://10.3.0.221>
.
Can anyone give a pointer?
Regards,
Paul
U 2005/04/05 21:06:29.459587 66.90.50.230:54115 <http://66.90.50.230:54115>->
10.3.0.221:5060 <http://10.3.0.221:5060>
ACK sip:8664365708@sipdev.mycompany.com SIP/2.0.
Via: SIP/2.0/UDP 172.16.1.42:5060 <http://172.16.1.42:5060>
;branch=z9hG4bK1177883239.
From: Paul Hazlett <sip:3212514240@sipdev.mycompany.com>;tag=1697105220.
To: <sip:8664365708@sipdev.mycompany.com>;tag=06922fcc.
Call-ID: 3209616113(a)172.16.1.42.
CSeq: 2301 ACK.
Content-Length: 0.
.
#
U 2005/04/05 21:06:29.692031 10.3.0.221:5060 <http://10.3.0.221:5060> ->
10.3.0.221:5060 <http://10.3.0.221:5060>
ACK sip:8664365708@sipdev.mycompany.com SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:10.3.0.221 <http://10.3.0.221>;ftag=1697105220;lr>.
Via: SIP/2.0/UDP 10.3.0.221 <http://10.3.0.221>;branch=0.
Via: SIP/2.0/UDP 172.16.1.42:5060 <http://172.16.1.42:5060>;received=
66.90.50.230 <http://66.90.50.230>;branch=z9hG4bK1177883239.
From: Paul Hazlett <sip:3212514240@sipdev.mycompany.com>;tag=1697105220.
To: <sip:8664365708@sipdev.mycompany.com>;tag=06922fcc.
Call-ID: 3209616113(a)172.16.1.42.
CSeq: 2301 ACK.
Content-Length: 0.
P-hint: Local Destination.
thx Steve,
actually i dont quite understand e ser.cfg my previous
project mate created.
here i added tis portion into my ser.cfg...can u teach
me where do i call t_on_failure?
if(method=="ACK" || method=="INVITE" || method=="BYE"
|| method=="REFER")
{
if(t_newtran())
{
t_reply("100","Trying -- just wait a minute
!");
if(method=="INVITE" || method=="REFER")
{
log("**************** vm start -
begin ******************\n");
if( uri =~ "conference" )
{
if(!vm("/tmp/am_fifo","conference"))
{
log("could not contact conference server\n");
t_reply("500","could not contact conference server");
};
}
else if( uri =~ "echo" )
{
if(!vm("/tmp/am_fifo","echo"))
{
log("could not
contact echo\n");
t_reply("500","could not contact echo");
};
}
else
{
if(!vm("/tmp/am_fifo","voicemail"))
{
log("could not contact
voicemail\n");
t_reply("500","could not contact voicemail");
};
};
log("**************** vm start - end
******************\n");
break;
};
if(method=="BYE")
{
log("**************** vm end/refer - begin
******************\n");
if(!vm("/tmp/am_fifo","bye"))
{
log("could not contact the media server\n");
t_reply("500","could
not contact the media server");
};
log("**************** vm end/refer - end
********************\n");
break;
};
}
else
{
log("could not create new transaction\n");
sl_send_reply("500","could not create new
transaction");
};
};
# Voicemail specific configuration - end
Pls do help me...
rgds,
Eric
Sure. Call t_on_failure("7") in route block 0, the
main route block
in
SER. This
will define an internal "flag" labeled 7. When the
current call "fails"
that is does not
get a 200 OK message SER will look for a failure block
with the label 7
and
execute the code therein. Note that 7 is an arbitrary
integer. You can
pick whatever
integer makes you happy and doesn't conflict with
existing flags.
In the failure_route most people simply re-write the
destination
hostname and
then t_relay the call to that new host. You may need
to re-write the
port number
too. In either case this host and/or port is where
your voicemail
application is
listening.
A failure_route is used instead of the regular route
statement
because
you want
this section of code to be reached when a status code
other than a 200
OK is
received for a call. An unanswered call gets something
like a 487 (you
should
check this for the actual number).
If you use SEMS you have yet another challenge. When
SEMS was first
deployed
it communicated with SER used a fifo. Since the fifo
is not a SIP
addressable
entity you needed a way for the failure_route to send
SIP messages to
SEMS.
Most people used a second instance of SER on the same
box but with a
stripped down config file. This second instance could
communicate with
SEMS
using a different named pipe.
Newer versions of SEMS do not have this limitation
but you need to
configure
the type of interface SEMS will use. fifo or socket. I
did one
installation this
way but then moved onto using Asterisk for voicemail.
Hope this helps.
_Steve
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I have some problem maybe some of you guys can help me with.
I have ATA device connected to hub -> router -> ser server (5060) -> asterisk (5061) -> pstn
sometimes when i am making a call the other side can not hear me, but i can hear the other side, it happens sometimes, and sometimes it's working fine.
where i can start to check the problem in order to solve it.
thanks for any help.
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Hi,
I am kind of new to the ser.
I installed, and run it. It is working well. I am happy
I tried to add Presenece module to Ser.
I did following
Modify ser.cfg
add=>loadmodule "usr/lib/ser/modules/pa.so"
add=>modparam("pa","default_expires",3600)
If I add follwing inside the route
add=>if(method==SUBSCRIBE){ handle_subscription("registar");}
Ser stoped. I can not start it
I do not know how to use this function. Please help me to solve this.
Thanks
Jaedong82
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Hi,
I'd like to know if there is an option on SER that allow voice traffic to
pass through the server after initial negotiation.
In my scenario, I want a communication between two softphones by means of
SER server, but I want that the traffic pass through it.
Is there an answer?
Best regards,
---------------------------------------
Alexandre Passito
Universidade Federal do Amazonas
Departamento de Ciência da Computação
LabVoIP - GRSD - UFAM - Brasil
---------------------------------------
Hi
Just come across a really strange problem, some of my REGISTER queries
do not hit the DB, even though they hit SER, anyone ever come across
this, it only seems to happen with a particular kind of hardware
(unbranded), and only started recently. No firmware upgrades done
Iqbal
I was running SER 0.9.0 with mediaproxy and the default mediaproxy
ser.cfg. I tested two xten clients behind two 2wire 180hg firewalls.
Everything works perfectly if both clients use port 5060 for signaling.
However, if you change either or both to a different signaling port,
such as having each client listen on port 11111, then SER inserts
incorrect entries into the via and record route sets. It will list
itself as sip:ser.whatever.com:11111 when you call a t_relay. The call
can set up fine, but you end up losing the bye's and cancels. Has anyone
else noticed this problem?
Thank you, SG
