Hello,
I want to make a call from PSTN to a SIP phone.
I use a cisco 5300 for answering/handling the PSTN number
and then forward the call to a SER box (0.8.14) .
POTS -> c5300 --> SER -> SIP softphone
My conf on c5300:
!
dial-peer voice 1003 voip
destination-pattern 12345
session protocol sipv2
session target sip-server
codec g711ulaw
!
sip-ua
sip-server ipv4:X.Y.Z.W
!
What is the appropriate configuration on ser.cfg in order to
handle this call and forward it to sip:userX@mydomain.com ??
Any hint about this ??
I've already done the opposite scenario (SIP->PSTN) :
ser.cfg:
if (uri =~ "^sip:2*") {
rewritehostport("1.2.3.4:5060");
forward(uri:host, uri:port);
break;
};
thanks for any help,
Kostas
--
how to solve this issue
i am root
ser is working but
#ps -A
3793 ? 00:00:00 ser <defunct>
--- Grigory Fishilevich <g.fishilevich(a)gmail.com>
wrote:
> hi,
>
> these messages have nothing to say. i think you have
> a permissions
> problem, you should take a look at /var/log/messages
>
> ? have you ser running or not?
>
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Does anyone know a free or lowcost billing solution which works with SER, before i've tried with some commercials ones but they do not worked with SER and they were Radius Supported ones so was creating its own database.
All i need is a few simple PHP files so i can add/remove users from there and a add table for rating depending on prefix and it should use SER's database which is already created. And do sample calculations for me, enter user id/loginname/callerid or whatever and show me caller/callee/duration/time/date/cost and under all a grand total with tax space. So simple but i do not know how to do in PHP. Even some examples or good URL's to read are fine for me.
I use SER 0.8.14 but may use 0.9.0 or newer if needed.
Thanks,
Ozan Blotter
After installing mysql 5.0.3 and running ser_mysql.sh
then
mysql> select * from user;
Returns:
ERROR 1146 (42S02): Table 'ser.user' doesn't exist
Any help is grealty appreciated
-kim
--
w8hdkim(a)gmail.com
hi list
any one know what could be the problem
here is the trace
i cannot use serctl moni
Error opening ser's FIFO /tmp/ser_fifo
Make sure you have line fifo=/tmp/ser_fifo in your
config
make_and_install_output.txt
--------------------------------
Makefile.rules:81: action.d: No such file or directory
Makefile.rules:81: crc.d: No such file or directory
Makefile.rules:81: daemonize.d: No such file or
directory
Makefile.rules:81: data_lump.d: No such file or
directory
Makefile.rules:81: data_lump_rpl.d: No such file or
directory
Makefile.rules:81: dprint.d: No such file or directory
Makefile.rules:81: dset.d: No such file or directory
Makefile.rules:81: error.d: No such file or directory
Makefile.rules:81: fifo_server.d: No such file or
directory
Makefile.rules:81: flags.d: No such file or directory
Makefile.rules:81: forward.d: No such file or
directory
Makefile.rules:81: hash_func.d: No such file or
directory
Makefile.rules:81: ip_addr.d: No such file or
directory
Makefile.rules:81: main.d: No such file or directory
Makefile.rules:81: md5.d: No such file or directory
Makefile.rules:81: md5utils.d: No such file or
directory
Makefile.rules:81: modparam.d: No such file or
directory
Makefile.rules:81: msg_translator.d: No such file or
directory
Makefile.rules:81: pass_fd.d: No such file or
directory
Makefile.rules:81: proxy.d: No such file or directory
Makefile.rules:81: qvalue.d: No such file or directory
Makefile.rules:81: re.d: No such file or directory
Makefile.rules:81: receive.d: No such file or
directory
Makefile.rules:81: resolve.d: No such file or
directory
Makefile.rules:81: route.d: No such file or directory
Makefile.rules:81: route_struct.d: No such file or
directory
Makefile.rules:81: script_cb.d: No such file or
directory
Makefile.rules:81: socket_info.d: No such file or directory
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Hello,
would anybody please tell me where can I get right version of uac module
for ser-0.9.0.
I downoaded cvs snapshot of ser-0.10.99
can I use uac of this snapshot on ser-0.9.0?
Mohammad
Gentlemen:
I saw a posting from Javarockx
http://lists.iptel.org/pipermail/serusers/2005-February/015687.html
mentioning that he wants to repost his reworked
ser.cfg. My search did not give me any positive
results. Can someone (Paul?) please email it to the
list?
Thanks,
Dave
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Can it do what I described?
Documentation isn't very good on that web site, but according to what I saw
I doubt it.
Also what does it mean SER based PBX? SER+voicemail?
Thanks,
Michael
On Friday 20 May 2005 07:20 pm, you wrote:
>
> we have an all in one ser solution that might be usefull for you
> http://www.wifi.com.ar/english/voip.html
>
> regards,
>
> ____________________________________________________________________________
>
> Jaime Garcia Ghirelli http://www.brujula.net
> jaime(a)fonosip.com http://www.fonosip.com
> http://www.wifi.com.ar
>
> ---------- Forwarded message ----------
> Date: Fri, 20 May 2005 15:49:35 -0400
> From: Michael Ulitskiy <mdu113(a)acedsl.com>
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Advice needed
>
> Hello,
>
> I'd like ask for advice on what is in your opinion the best solution
> in the following scenario.
> I have a bunch of sip servers (asterisk boxes as my users need pbx
> functionality) that can make sip call to each other and my PSTN
> gateway. Now I want to purchase PSTN terminitaion in several
> different markets (and probably more in the future). All those
> terminations will require authentication.
> I want all my boxes when they see non-local call to send it to a
> central routing server that would determine where this call should
> be sent and authenticate to the appropriate provider so that I don't
> have to configure all credentials on all asterisk boxes. Also I want
> it not to deal with the media at all. All media streams should go directly
> from asterisk box to the PSTN termination provider.
> So basically it should be central SIP router that is able to authenticate
> calls if neccessary.
> I thought I could do it with SER and its UAC module, but it appears
> UAC module doesn't work and probably won't work (see my previous
> post in this list about UAC backport to 0.9.0).
> Also I don't want to use asterisk in this place as asterisk always wants to
> stay in media path and I'd really like to avoid of getting into hassle with
> re-invites.
> So the question is what are my options and what you would advice
> as a solution. Are there any software out there that can do it (preferably
> open-source, of course) or what else you could suggest to do to get
> desired results.
> Thanks a lot,
> --
> See you later,
> Michael
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
--
See you later,
Michael