Hello
I'm trying to implement the following scheme
<User LAN, ATA186, PC, etc> --- < Linksys router with dynamic public IP > -- shdsl ---
--- ISP --- .... --- VoIP provider network ( SER, PSTN gateway, etc )
It will be nice, of anyone could give me an advice how to implement
the possibility of calls to and from VoIP devices INSIDE User network on
private IPs to the VoIP provider network on fixed static IPs.
Maybe I misunderstood something, but it's not clear to me, how the
incoming call to user will be possible in this scenario.
Linksys router = NAT + DHCP + 1-8 port hub
Thanks.
--
Michael Vasilenko
I'm trying to get SER to work properly with the call forwarding feature of
an ATA-186. The ATA-186 forwards calls (#90number#) by doing a BYE with
an "also" header. SER doesn't seem to do anything with this at all, and
the call just sits in limbo until everybody hangs up.
Call forwarding on busy/no-answer works, but I think that's because the
ATA is doing a refer instead.
Any clue where I should be looking for this? Is SER supposed to support
this BYE/also method, and I have something misconfigured, or is there
additional work that needs to be done to support this?
Thanks!
- Mike
Hi,
we are suffering the same problem. The serctl script was originally written
for Linux and is a little incompatible with other systems. Even if you
change the shell from sh to bash there are still some utilities like "tail"
which has a diffirent syntax than from Linux.
I am porting this script to solaris for our systems where no bash is
available. If this is also useful for you I can send you later.
Regards
Yang
On Apr 02, 2003 at 08:26, Steve Blair <blairs(a)isc.upenn.edu> wrote:
>
> Hello:
>
> I'm just getting started with my implementation of
> SER on FreeBSD 4.7-RELEASE. I've read the
> documentation, installed Apache and mySQL and
> would like to add users for my domain.
>
> I've tried adding an administrative user using serctl
> however this script fails for reason I cannot explain.
> Here is what I did:
>
>
> serctl add user1 password1 email1(a)mydomain.com
> read: Illegal option -s
>
> read: Illegal option -s
> Try changing the first line of the script form #!/bin/sh to #!/bin/bash
> (the read in sh does not support the -s option).
> BTW: this is fixed on CVS for the new version (but don't try the CVS
> code until next week, we're commiting a lot of changes right now).
>
> Andrei
hi all,
i have installed SER (ver 0.8.10) with MySql support.
i have a problem. in ser.cfg when i set the "fork" parameter to "no", SER
is not coming up at all. if the default "yes" is used, the server comes up
just fine.
pls help. i want to disable forking somehow.
rgds,
sunithi
since many people mentioned that they are using cisco 5x00 series
sip/pstn gateways, it would be nice if also others than me would push
cisco to implement digest authentication in its ios software. currently
5x00 namely is the only sip UA that i'm aware of that doesn't support
digest (or any other) authentication of sip requests.
another annoying thing with 5x00 is that i cannot configure the host
part of its from uri. it always uses as the host part the numeric ip
address of the interface via which it is sending out the sip request.
-- juha
Hi Tomas,
we have an AS 5200 instead of AS 5300. Can we use it as SIP2PSTN gateway too?
Which IOS are you using? Must the IOS contain the MCM?
Best Regards
Yang
hi,
I note that there is a example of click_to_dial in the ser/examples/web_im directory, but it seems not to be implemented fully. I want to know how to implement click_to_dial in ser,does that need RTP stack and other things?
thanks.
　　　　　　　　Joe_chen
　　　　　　　　chenhb(a)sict.ac.cn
　　　　　　　　　　2003-04-03
It is possibly a .net only option. The Messenger help seems
to indicate that it should work for any communications service,
but the help/document files are pretty thin.
I guess I need to see if I can find another SIP capable IM
interface.
Dan
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Wednesday, April 02, 2003 12:32 PM
To: Dan Austin; serusers(a)lists.iptel.org
Subject: RE: [Serusers] Re: windows messenger
At 10:05 PM 4/2/2003, Dan Austin wrote:
>Calls to the PSTN work for me as well, but I've run into a small snag
>using the IM functions.
>
>Messenger claims I can added multiple parties to the IM session, but
>the menu option is not available. Am I running into a Messenger issue,
>or a SER limitation?
Maybe it is just a .net option? It seems unlikely to me SER is guilty.
-jiri
Hi,
Installed ser (0.8.10 src) and mysql db. Tested the
register using some diffrent UAC. one of UAC got the
problem. When the UAC sent the register msg to ser,
ser generated more than 1M msg in log file (using log
level 8). A few seconds later, UAC got "513 msg too
big". I used the defualt max_len and the msg is kind
samll. see the attached register msg.
Thanks,
Alan
-----------------------------------------------
REGISTER sip:191.191.29.58:25060 SIP/2.0
Via: SIP/2.0/UDP 191.191.29.62:5060
From:
3001<sip:3001@iptel.org:5060;user=phone>;tag=27b74-12ec
To: 3001<sip:3001@iptel.org:5060;user=phone>
Call-ID:
a8cce363000000000000000000000000(a)191.191.29.62
CSeq: 101 REGISTER
Max-Forwards: 70
Expires: 900
Contact: <sip:3001@191.191.29.62:5060;user=phone>
Supported: timer
Content-Length: 0
=====
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