Hi all,
I am having problem in creating MySQL database, I followed the given below
instruction
create MySQL tables
- if you have a previously installed OpenSER on your system, use
/usr/sbin/openser_mysql.sh reinstall
to convert your OpenSER database into new structures
- otherwise, if this is your very first installation, use
/usr/sbin/openser_mysql.sh create
to create OpenSER database structures
(you will be prompted for password of MySql "root" user)
but when i tried to create it, this is what happened
/sbin/openser_mysql.sh create
MySql password for root:
Domain (realm) for the default user 'admin': admin
/sbin/openser_mysql.sh: line 201: openser_gen_ha1: command not found
HA1 calculation failed
any help regarding this would be favourable
thanks
naveen
Arrgh!
Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing?
How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore.
t_replicate("192.168.10.7","5060");
t_replicate("192.168.10.8","5060");
t_replicate("192.168.10.200","5060");
Doug.
Sending to the list as more people might be interested...
I read offer/answer RFC quite a long ago but as far as I remember
there is no limit in the parameters you can change in a new
offer/answer exchange.
According to this, UA1 can change the port and the media direccion in
the same offer/answer exchange.
Samuel.
2005/12/15, tcchan <tikchoong.chan(a)redtone.com>:
> Dear Samuel,
>
>
> Thanks you very much for your answer.
>
>
> I would like to consult you the following:
>
> normal call,
>
> UA1 ----- invite --port 30000-------> UA2
>
> UA2 <---------OK port 20000 ---------UA2
>
>
> UA1 hold the call,
>
> UA1 -----invite --port 30000(hold) ----> UA2
>
> UA1 <--------OK port 20000 -------------UA2
>
>
>
>
>
> UA1 unhold the call,
>
> UA1 ------invite -----port 30001 -------> UA2
> UA1 <----------OK port 20000 -----------UA2
>
>
>
> Noticed that when UA1 unhold the call, it uses a different port.
> Is UA1 SIP compliant.
>
> Thanks again and Regards,
>
> TC Chan
>
>
> On Wed, 2005-12-14 at 09:18 +0100, samuel wrote:
> > UA is not SIP compliant. It must include the SDP in the OK.
> >
> > Samuel.
> >
> > 2005/12/14, tcchan <tikchoong.chan(a)redtone.com>:
> > > Dear All,
> > >
> > > I encountered a UA that behave as follow:
> > >
> > > UA1 call UA2
> > >
> > > UA1 ---- invite with SDP -----> UA2
> > >
> > > UA1 <------OK with SDP --------UA2
> > >
> > >
> > > UA1 hold the call,
> > >
> > > UA1 ------invite (hold) ------>UA2
> > > UA1 <--------OK but not SDP ---UA2
> > >
> > >
> > > UA1 unhold the call
> > >
> > > UA1 ------invite ----------->UA2
> > > UA1 <-------OK but also no SDP---UA2
> > >
> > >
> > >
> > > I found that most ATA(UA) would attached a SDP when reply an OK whether
> > > the call is hold or unhold. But there is one ATA that does not (UA2).
> > >
> > >
> > > I noticed that some ATA--Linksys (UA1) would send a BYE to UA2 if it
> > > receive a OK with no SDP in the hold case.
> > >
> > > My Question is :
> > >
> > > Is UA2 behave according to RFC?
> > >
> > >
> > >
> > > Would deeply appreciate if someone could let me know the answer as I
> > > always have trouble reading RFCs...
> > >
> > >
> > > Thanks fot any help available.
> > >
> > > Regards,
> > >
> > > TC Chan
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> >
>
>
>
>
Hello guys,
I read in the documentation for the ACC module that it supports multi
call-leg accounting via the use of AVPs, which comes in handing when doing
call forwarding. However, I've googled the way with all the keywords that I
could think of and I cant find an example of this feature. I understand that
you have to turn it on using:
modparam("acc", "multi_leg_enabled", 1)
modparam("acc", "src_leg_avp_id", 110)
modparam("acc", "dst_leg_avp_id", 111)
But it doesn't show or say when/how to use the AVPs. Can anybody provide a
sample snippet or their openser.cfg file to demonstrate how it's used?
Thanks in advance!
Lenir
P.s. perhaps put a sample config in the documentation. The one there doesn't
show how to use Multi Call-leg billing.
Well Ken
gafachi service has to offer 'refer' method. No other way out. I am working
with Primus and you can take minutes from them. They offer it.
>From: Ken Rozinsky <xraycharlie(a)gmail.com>
>To: Kapil Dhawan <sersavvy(a)hotmail.com>
>Subject: Re: [Serusers] simple click-to-call
>Date: Wed, 14 Dec 2005 08:49:55 -0700
>
>Thanks for the help,
>
>I have it set up to the point where it will call me back but the
>gafachi service does not support refer :(
>
>I'm looking around for a similar service that does. Know of a good one?
>
>Also, I've been trying to work around the refer by sending another
>invite. The other phone will ring now after I pick up my phone but I
>hear no ringing on my phone. The lines are also not joined when the
>other party picks up.
>
>Any thoughts on how to do this using invites?
>
>Regards
>
>Ken
>
>
>
>On 12/13/05, Kapil Dhawan <sersavvy(a)hotmail.com> wrote:
> > If SER is running, that means fifo is ON.
> >
> >
> > >From: Ken Rozinsky <xraycharlie(a)gmail.com>
> > >To: Kapil Dhawan <sersavvy(a)hotmail.com>
> > >Subject: Re: [Serusers] simple click-to-call
> > >Date: Tue, 13 Dec 2005 11:24:58 -0700
> > >
> > >Thanks for the info.
> > >
> > >A question for you if you have the time:
> > >
> > >The script checks if it can write to /tmp/ser_fifo and fails.
> > >The comment mention having to have the fifo server running. How do I
> > >check if this is running or not?
> > >
> > >I see in the documantation a mention of fifo_db but I don;t think I
> > >need this as am not worried about persistance.
> > >
> > >The permissions on /tmp/ser_fifo are currently prw-rw----
> > >
> > >I tried running the script as root and get the write error also.
> > >
> > >Thanks again.
> > >
> > >Regards
> > >
> > >Ken
> > >
> > >
> > >On 12/12/05, Kapil Dhawan <sersavvy(a)hotmail.com> wrote:
> > > > Ken
> > > >
> > > > Click-to-dial is present in example directoy as ctd.sh. It takes two
> > > > arguments From and To. To run it from Web-interface, you need to run
> > >this
> > > > command thru script.
> > > >
> > > > Regards
> > > >
> > > >
> > > > >From: Ken Rozinsky <xraycharlie(a)gmail.com>
> > > > >To: serusers(a)lists.iptel.org
> > > > >Subject: [Serusers] simple click-to-call
> > > > >Date: Mon, 12 Dec 2005 17:27:18 -0700
> > > > >
> > > > >Hello,
> > > > >
> > > > >As an exercise in learning how all this stuff works, I'm trying to
>set
> > > > >up a simple click-to-call service for our office to use. And
> > > > >potentialy to allow clients to use vai the web site to save them on
> > > > >the long distance charges.
> > > > >
> > > > >I set up Asterisk with the TACI script and got it to all work with
>an
> > > > >account on gafachi but I noticed that it could only handle one call
>at
> > > > >a time.
> > > > >
> > > > >I was told that openSer would allow more than one call at a time so
> > > > >I've installed it (basic no mySql) and have it up and running.
> > > > >
> > > > >Now, being new to this I have no idea how to connct to gafachi and
>do
> > > > >a click-to-call type thing.
> > > > >
> > > > >Would anybody be able to help me with this?
> > > > >
> > > > >Regards
> > > > >
> > > > >_______________________________________________
> > > > >Serusers mailing list
> > > > >serusers(a)lists.iptel.org
> > > > >http://lists.iptel.org/mailman/listinfo/serusers
> > > >
> > > > _________________________________________________________________
> > > > Shah Rukh fan? Know all about the Baadshah of Bollywood. On MSN
>Search
> > > > http://server1.msn.co.in/profile/shahrukh.asp
> > > >
> > > >
> >
> > _________________________________________________________________
> > How good are you in a Formula One car? Play now
> > http://server1.msn.co.in/sp05/tataracing/onlinegame.asp
> >
> >
_________________________________________________________________
Shah Rukh fan? Know all about the Baadshah of Bollywood. On MSN Search
http://server1.msn.co.in/profile/shahrukh.asp
Normally I would call a statement like:
If (uri=^sip:123456@somewhere.com)
{
..do something
}
I would like to implement a dial plan using the "From:" header.
i.e.:
If (from_uri=^sip:123456@somewhere.com)
{
..do something
}
Anyone know if something like this exists?
Leo P.
Hi,
Is there a way or workaround to generate AVPs with header
values as AVP name? It seems to me that the current AVP
concept defines the AVPs at OpenSER startup, so
header-value-based names are not feasible.
What I need is the following: Store/retrieve a (key,value)
pair where the key is the current message's call-id and
the value some combination of header values and pseudo-
variables.
e.g. use the ops
avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts");
avp_pushto("$Myheader","$hdr(call-id)");
in order to append the following header to the SIP message:
Myheader: 23459(a)10.0.0.1-4235627623
Any idea how to do this in OpenSER?
thanks in advance
--Joachim
Please disregard my previous posts.
Apparently I solved the problem not by replacing the 5060 port but by adding
the new port at the ser.cfg
listen=10.0.1.99
port=5060
listen=10.0.1.99
port=8080
I setup the listening port and the SIP port on the Xlite to 8080 and is
registering fine.
ngrep is showing that the source port and the destination port both ways is
8080 so it should be working.
I will test it in the real world and inform.
Thanks and regards
Juan
Samuel,
It seems that t_replicate() requires a host and port as parameter. It looks like it doesn't use what's in the destination set to deliver messages... which breaks stuff. Arrgh, I'm going in circles!
-----Original Message-----
From: samuel [mailto:samu60@gmail.com]
Sent: Wednesday, December 14, 2005 10:27 AM
To: Douglas Garstang
Cc: users(a)openser.org
Subject: Re: [Users] Tricky Question
Have you tried t_replicate instead of t_relay???
This t_replicate function should send the REGISTER to the other
registrar and absorve the answers. However this t_replicate has (or
had??) an inconvenient: it can be used with only one secondary
host...so it's not possible to use it with several asterisks. Please,
correct me if I am wrong.
Samuel.
2005/12/14, Douglas Garstang <dgarstang(a)oneeighty.com>:
> Here's a tricky question for the list.
>
> I have phones registering with OpenSER. After authenticating and saving the location, it forwards each registration request onto multiple Asterisk systems who send back Trying, OK, etc. OpenSER forwards these back to the phones eventhough they don't need to be. I want to stop them from getting back to the phones, because from the phone's perspective they've already registered, and now these extra Trying, Ok messages are coming back. Furtunately it doesn't seem to be causing any problems for the Polycom phones, but I'd still like ton get rid of them.
>
> How....? Maybe call setflag() when I get a registration, and then look for messages coming back with the flag?
>
> Doug
>
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>