Dear Madam/Sir,
We are a data communication provider in Egypt, and we are very much
interested in deploying SER.
We are particularly interested in having a voicemail feature as one of
our offerings, and we have tried the CVS (8.11) version but we have
noticed (naturally) that it's not stable yet.
Could you please give us a rough estimate on when you believe a beta or
alpha release will be available?
Thanks a lot for your time and cooperation.
Regards,
Ahmed Abdoulla
Quality Unit Manager
Acting WebDev Unit Manager
TE Data
11, Cleopatra St, Heliopolis
TEL : +20-2-4166660
FAX : +20-2-4166700
Hi,
I would like to use the registrar module, but ser crahses after calling the
save() or lookup() functions. Any idea?
syslog:
BUG: tcp_main_loop: dead child 6
tcp_main_loop: read response= bffff638, 2003 from 6 (20531)
BUG: tcp_main_loop: unknown cmd 2003
child process 20531 exited by a signal 11
core was not generated
INFO: terminating due to SIGCHLD
INFO: signal 15 received
Thanks,
--
---------------------------------------------------------------------------
Miklos Tirpak
Computer and Automation Research Institute e-mail : mtirpak(a)sztaki.hu
of the Hungarian Academy of Sciences phone : (361) 279-6011
H-1132. Budapest, Victor Hugo u 18-22 fax : (361) 279-6021
Hello
Is there a built in function that can check to see if a user exists (not
registered). I have a need to send different error codes depending on
whether a phone number is unallocated or it is allocated but not
registered. What I require is this
if (!lookup("location")) {
if (user does not exist) {
sl_send_reply ("604", "Does not exist anywhere");
break;
}
else {
sl_send_reply ("404", "Not Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
Thanks
Luke
during my pstn gateway survey, some people asked about cheap PSTN
gateways. Let me share what I know, speak up if you know more.
analog
-------
AudioCodes and AlliedTelesyn provide SIP FXO gateways wuith four ports
for about $1000. We have both of them in our lab (i.e., they do exist)
but haven't tested them yet (i.e., we don't know if they work).
MediaTrix might have some too, perhaps sightly cheaper, but they denied
shipping to us as we are not in their interoperability club (which makes
me cautios about their interoperability).
arrayvox was claiming to have some very cheap ones, but I neither know if
they shop nor if they work. The same for welltech.
in general, not only is a challenge to make the sip part of it work, but
also the analog -- tone recoginition may be pretty challenging. It differs
country from country and pbx from pbx.
digital
-------
I don't know -- some people meant Asterisk would do that, some other meant
not really.
other
-----
someone told me vegstream (both digital and analog) was good and afforable.
more can be found at
http://www.iptel.org/info/products/index.php?category=gateway&name=Gateways…
--
Jiri Kuthan http://iptel.org/~jiri/
hello every one,
I am facing some problem in the OPTIONS method.
I am getting "404 not found" response.
I wanted to query the server for its capabilities.
I wanted to know which are the methods it supports from the 200OK response of my OPTIONS method.
My packet was as below
OPTIONS sip:192.168.1.9 SIP/2.0
Via: sip/2.0/UDP 192.168.1.9:5060
Max-Forwards: 70
To: <sip:192.168.1.9>
From: <sip:geetha@192.168.1.21>;tag=1047012665620
Call-Id: 82963748691a049dc5aadf81b162c894(a)192.168.1.9
CSeq: 1 OPTIONS
Contact: <sip:192.168.1.9:5060>
Accept: application/sdp
Content-Length: 0
Can some one tell me How SER works for OPTIONS method and why am i getting that "404 not found" response.Is there some problem with my packet?
Thanks
Geetha
--
********************************************************************
eRiva provides IT Solutions & Services to companies worldwide.
Our State of the Art Research & Development Centers provides
Innovative Solutions to global customers.
********************************************************************
--
At 05:47 AM 3/29/2003, Dan Austin wrote:
>Can anyone recommend a SIP stack for Linux, where the code is documented well
>enough so that someone not an expert on SIP programming can
>figure it out?
A possibility is our FIFO UA, part of SER. It was built to couple SIP-unaware
applications with SER over stdin/stdout interface. We use it for our click-to-dial
and voicemail application. The adventage is it is pretty simply and you don't need
to run into stack details. For sake of completeness: some in our team would prefer
a stand-alone SIP stack better, but we don't have any yet.
If that is interesting for you, we will be glad to provide you with guidance.
-Jiri
hi,
Thanks for the quick reply.
Geetha
---------- Original Message ----------------------------------
>From: Nils Ohlmeier <nils(a)iptel.org>
>Reply-To: seruser(a)iptel.org
>Date: Sat, 29 Mar 2003 05:08:36 +0100
>
>>On Saturday 29 March 2003 05:01, Nils Ohlmeier wrote:
>>> On Saturday 29 March 2003 04:49, Geetha Shree wrote:
>>> > Can some one tell me How SER works for OPTIONS method and why am i
>>> > getting that "404 not found" response.Is there some problem with my
>>> > packet?
>>>
>>> the answer is pretty simple: Ser is currently not yet able to answer
>>> OPTIONS to the server itself. Maybe it is possible to create a correct
>>> answer with a lot of script commands (e.g. if==OPTIONS and user not present
>>> in uri { sl_reply(200); append_hf(header filed with the methods);break}),
>>> but their is support for this built into Ser or one of its modules.
>> ^^^
>> Errata: is no support
>>
>>> As a proxy it tries to find the user, to which it should process the
>>> request, which is not present, and because of this you get a 404 back.
>>
>>Greetings
>> Nils
>>
>>
>
>--
>********************************************************************
>eRiva provides IT Solutions & Services to companies worldwide.
>Our State of the Art Research & Development Centers provides
>Innovative Solutions to global customers.
>********************************************************************
>
>
>
>--
>
--
********************************************************************
eRiva provides IT Solutions & Services to companies worldwide.
Our State of the Art Research & Development Centers provides
Innovative Solutions to global customers.
********************************************************************
--
I'd guess that many folks on the list are as interested in
finding a low cost PC based gateway as I am.
My research has uncovered on possible candidate, and caused
some confusion at the same time.
I came upon a product called Wildcard T100P. A single loop
T1/PRI card with linux drivers, for the oh-so-reasonable price
of US $500. The catch seems to be that there is no gateway
code written to date to use the card, at least no code whose
only function is to be a gateway. There is also a single
port FXO card for US $100
It turns out that the OSS PBX Asterisk can leverage the card,
has support for SIP, and is in general really cool, but too
feature rich to be used as a PSTN gateway. Sure the price is
right (free).
Has anyone worked on or with the Wildcard? Details about the
card can be found at http://www.zapatatelephony.org/
What is lacking appears to be a package that can leverage
the Wildcard library with a SIP frontend. I am tempted to
try to workout what the framework of such a package would
look like, and try to rough out some code. The problem with
that is I'm no coder*. The source for the Wildcard driver is
well documented, so I may try anyways. Can anyone recommend
a SIP stack for Linux, where the code is documented well
enough so that someone not an expert on SIP programming can
figure it out?
* Not professionally. I have needed to debug C source before,
and have written small tools that to be honest were not at all
elegant.
Dan
Juha's ENUM lookup routine currently looks up the e164.arpa tree for a
phone number and searches for a sip record, which is returned and used to
rewrite the current uri.
ENUM also allows tel: records amongst others.
What, if anything should be done with those, and how should they be treated
relative to sip records?
I would assume a tel: record could be interpreted as a kind of call
forward, which could be arbitrarily staged.
One way of handling this could be to explicitely look for tel records with,
e.g., an argument to enum_query() to have any tel records rewrite the
current uri, and then go on to look up the result (maybe recursively which
requires a loop counter and an abort condition).
so the resolution could look like
while (loopcount--) {
enum_query("tel");
enum_query("sip");
}
what do you think?
-Michael
hi,
thanks. what i am looking for is not a full fledged accounting module. i
need to know if currently generated call records can be put into MySql
database. if yes, is there any tool available to view these records?
rgds,
sunithi
-----Original Message-----
From: jh(a)lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
Sent: Thursday, March 27, 2003 8:21 PM
To: Sunithi JP
Cc: 'serusers(a)lists.iptel.org'
Subject: RE: [Serusers] help with accounting using mysql
Sunithi JP writes:
> thanks for the quick response. cud u pls tell me if accounting thru
RADIUS
> is working?
it depends how you define "working". the problem currently is that
there is no easy way to know if a bye, for example, was sent by a local
or foreign user. before that is information is available, any
accounting is difficult to make to "work".
if you mean a version of radius accounting that doesn't crash and
produces "records", i'm sure that such a module will be included in the
next release.
-- juha
