I am looking for a software for challanging the load limits of preinstalled SER server. There are both free (i.e. sip bomber) and priced (i.e. winsip) products. Free products are installed on Linux platforms, but the others can be setup on windows.
If you know such programs, please inform me about the test results and your experience on those products. Any other software will be beneficial for me..
Thanks,
Dev
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I want to allow users to have multiple numbers that they can be found
at. For example, I want to have SER ring my SIP phone first, then my
cell phone, then department admin, and finally my voicemail. However,
a different user may want to first ring his sip phone, then Dave's SIP
phone and then to voice mail.
I created a custom table called "followme" and in it I have two
columns, username and newusername. In my failure_route[1] I call
exec_dset("junk(a)junk.com;/etc/ser/follow $USER") and it will return
the correct value. This works pretty well as long as the returned
value is an external number that I pass off to the PBX. However, I
want to be able to do one of two things. 1) I want to run the
returned value through Lookup(location) 2) I want to have condition
statements based on the returned URI. For example I want to have: if
(uri =~"^xyz@)
I think the issue is that the exec_dset really does an append branch,
so that the original URI is not changed. I already have found taht I
need the junk(a)junk.com in my exec_dset command so taht SER recongizes
a new branch. But how can I work with that branch?
I hope this makes sense....
Aaron
Hi,
I think that the ACK after INVITE is only the callee(UAC) receives my
INVITE and ready to start RINGING. But this time is not the really
Call Start.
Generally, the start time of call should be callee hangup the phone
after ringing.
Do you have any idea about it?
On 4/28/05, info(a)beeplove.com <info(a)beeplove.com> wrote:
>
> Actually, to get call duration, you need to use ACK and BYE (both should
> have status 200).
> INVITE sends an request to other party to initiate a call.
> Other party send ACK with status 200 to receive the call.
>
> Original Message:
> -----------------
> From: Charles Wang lazy.charles(a)gmail.com
> Date: Thu, 28 Apr 2005 02:49:42 +0800
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] HELP: how to get correct call duration??
>
>
> Hi, ALL:
>
> I know ser can insert INVITE & BYE records to acc table.
> Maybe the timestamp of BYE record is correct but the timestamp
> of INVITE is not equal to the call start(after the callee answered).
> Is there any method to generate a record with timestamp when the callee
> answer?
> I think that it should be the callee responses "200 OK" after "100 Trying".
> How to do it???
>
> --
>
> Best Regards
> Charles
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
> --------------------------------------------------------------------
> mail2web - Check your email from the web at
> http://mail2web.com/ .
>
>
--
Best Regards
Charles
Hi all,
I'd like to know if there's some way i can have a list of authorized UAs i am
willing to deal with -rejecting all others- using SER.
I have noticed that there are some broken UAs like X-Pro v2 and surely some
others, that don't even send the BYE and there's no way i can have reliable
accounting with those UAs.
If anyone has found a way to do this, or has some patch, please lemme know.
Thanks in advance!
Enrique Vadillo.
Dear All,
After a month and over 200 downloads, we have updated the Getting Started
documentation.
We now cover in-depth description of supporting PSTN gateways as well as
updating some bugs that we had previously found.
So please download the new version 4 ! ! <http://www.ONsip.org>
www.ONsip.org
Feedback we have so far received is very positive and we hope that by
sharing with you our own experiences we are making the SER community wider
and more knowledgeable.
Regards
Paul, Simon, Greger
Hello,
I'm trying to comprehend loose routing concept and I have
a question that concerns me.
As far as I understand loose routing says that if there're Route
headers in a message it should be forwarded according to the URIs
set in Route headers.
I thought that this is true only within a dialog, but RFC3261 (part 16.6) says:
"Requests establishing a dialog may contain a preloaded Route header field."
Also SER manual says: " the failure not to include loose routing in your scripts
may lead to infinite loops. Make sure that you include the following script
fragment immediately after request sanity checks" and provide the following
piece of code:
if (loose_route()) {
t_relay();
break;
};
which as far as I understand unconditionally forwards message if Route header
is present.
So I'm wondering what about security? If I follow this guidelines how I would
shield my PSTN gateway if anyone can construct message and
pre-load it with URI of my gateway and all my proxies must honor it.
For example I have a PSTN gateway on ip address 10.1.1.5 and proxy
on 10.1.1.10 that supposed to interface outside world.
So I guess if someone construct a message like this:
INVITE sip:12345@somewhere.com SIP/2.0
...
Route: <sip:12345@10.1.1.5;lr>
my proxy will forward it to PSTN gateway and it will make outbound call.
Is this true? Please enlighten me on this.
Thank you,
Michael
Hi,
Is anyone using permissions Module? I read docu from web and I have no idea
about format of permission&deby files.
Would anyone kindly provide this format?
Great thanks.
Leon Sun
Michael.
By "tar" you mean the rpm installation?. If so i'm not sure what to
do, i'm sending this mail to the mail list to see if someone can give you a
hand.
Regards,
Ricardo.-
> -----Mensaje original-----
> De: michael p [mailto:mikep3000@hotmail.com]
> Enviado el: Martes, 03 de Mayo de 2005 12:17
> Para: rmartinez(a)redvoiss.net
> Asunto: RE: [serusers] mysql.so
>
>
> thanks but i didn't install ser with the source but with the tar
>
> so what can i do?
>
> thanks
>
> regards
>
> M.
>
> >From: Ricardo Martinez <rmartinez(a)redvoiss.net>
> >To: 'michael p' <mikep3000(a)hotmail.com>, serusers(a)lists.iptel.org
> >Subject: RE: [serusers] mysql.so
> >Date: Tue, 3 May 2005 12:16:37 -0400
> >
> >Hello
> >You must re-compile you ser, but this time make sure you
> edit the Makefile
> >file in the line "exclude_modules?=" and remove mysql module.
> >That should work.
> >
> >Regards,
> >Ricardo.-
> >
> >
> > > -----Mensaje original-----
> > > De: michael p [mailto:mikep3000@hotmail.com]
> > > Enviado el: Martes, 03 de Mayo de 2005 12:10
> > > Para: serusers(a)lists.iptel.org
> > > Asunto: [serusers] mysql.so
> > >
> > >
> > > hi
> > >
> > > i don't find the modules mysql.so that has to be in
> > > /usr/lib/ser/modules
> > >
> > > i don't know it's not there
> > >
> > > i can't modify the file /etc/ser/ser.cfg cause this module
> > > doesn't exist
> > >
> > > anyone knows why?
> > >
> > > where can i find it?
> > >
> > > thanks
> > >
> > > M.
> > >
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
>
>
I appreciate this has probably been asked time and time before, but I can
not find any documentation on it which works in my situation. I have
tried nathelper.cfg, rtp.cfg, al.cfg but to no avail. I have even tried
home grown cfg's
|---------sipsak for external registration
|
| rtpproxy
\|/ +
sip server <-> ser <-> asterisk
^ ^
| |
\|/ \|/
sip phone mgcp/iax phone
The ser and asterisk programs are running on the same box. This box has
an external and internal interface. ser listens to 5060, asterisk sip
listens to 5061. Internal phones connect to the internal interface of
this box. I used fix_nated_contact to modify the contact header, however,
looking at the xlog output this does not appear to be happenning.
/usr/local/sbin/ser[7077]: time [Tue May 3 17:04:33 2005] method
<REGISTER> r-uri <iptel.tgfslp.dalmany.co.uk> src_ip <192.168.4.5> contact
header <Line A <sip:5561@192.168.4.5:5061>;expires=3600>
Does anyone have any suggestions on how to fix this problem?
Unfortunately, I need this setup to proxy rtp through the box.
Many thanks,
Spencer
hi everybody,
anyone knows where is there a tutorial about serctl?
i don't understand how to use it?
i'm trying to enter some lines but when i do enter it doesn't happen things.
that stays on serctl
thanks
