Hi all,
Could any one help me to understand the back-up servers concept.
In the RFC 3263, section 2 says the following
" It is possible for elements to fail in the middle of a transaction.
For example, after proxy 2 forwards the request to UA 2, proxy 1
fails. UA 2 sends its response to proxy 2, which tries to forward it
to proxy 1, which is no longer available. The second aspect of the
flow in the introduction for which DNS is needed, is for proxy 2 to
identify a backup for proxy 1 that it can send the response to."
This talks about the Proxy 2's work ,but how the backup server will
collect the Details of the calls handled by the Proxy 1?.
Thanks in advance
John Miller
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Hello,
Please, is anybody able to provide the complete list of RFCs and drafts
SER is compliant to:
- in the current official release 8.14
- in the source archive ?
Thank you for your help
It was a permission problem. Thanks
but now I have another problem when I click on email link to confirm a
subdcription I get this error:
error in Sql query, line 43.
It is about confirmation.php but I didnt find anything strange.
Any help?
thanks
----- Original Message -----
From: "Terry Wilson" <otherwiseguy(a)gmail.com>
To: "Rosario Pingaro" <rpingar(a)italycom.it>
Sent: Thursday, February 10, 2005 7:19 PM
Subject: Re: [Serusers] Warning: fopen(/tmp/ser_fifo):
>> When I access the serweb I get this error:
>>
>> Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied
>> in
>> /usr/local/apache2/htdocs/serweb/functions.php on line 206
>>
>> Howto fix it?
>
> Try adding 'fifo_mode=0666' to your ser.cfg file and restarting ser.
>
Hi All.
I'm using ser-0.9 with mediaproxy-1.2.0
We've been trying to track down a problem whereby different ATAs stop
working either partially or crash entirely.
We have a UTstarcom iAN-02EX with firmware 1.8.2.15a (released a few weeks ago).
When I telnet to the ATA and enable debug messages I see this:
Not a valid SIP request!
Error: Parse error...
The above error is generated everytime SER sends a UDP keep-alive
packet shown below:
23:35:21.494441 pcp02356144pcs.clbrtn01.fl.comcast.net.5060 >
ssvsoho137.smartcity.net.58317: udp 4 (DF) [tos 0x10]
0x0000 4510 0020 69c2 4000 4011 24ee 4454 f278 E...i.@.@.$.DT.x
0x0010 425a 32e6 13c4 e3cd 000c 5c37 0000 0000 BZ2.......\7....
Can anyone tell me if this is a SER issue or not?
Regards,
Paul
Hello
it help that version I have to use so that operate correctly SER+NAT+RTPPROXY
SER version =?
Nathelper version =?
Rtpproxy version=?
thanks
Saludos.
A su Disposición.
Sergio Gerardo Festa
Dto. Técnica
TEL 011-52190654/52/53 // 4334-1014
24HS 011-15-5337-9684
How large are the sip messages? Do you have an SBC deployed at one/both
locations?
I've observed that SER will truncate message that it forwards if they
become too large. This is usually happens with two clients behind an SBC
when large via parameter have been inserted to maintain SBC state within
the sip dialog. Both Ingate and Jasomi products do this for example. The
result of these large via's is that the message become too large and SER
tend to truncate the SDP portion of the message. The connection
generally sets up with regard to signaling, but the media path always
fails.
I have observed this behavior while making a sip to sip call between two
xten clients, both positioned behind an ingate separator. Both clients
were registered to the same SER with RtpProxy (Redhat Enterprise AS
3.0).
-----Original Message-----
From: Andrei Pelinescu-Onciul
[mailto:pelinescu-onciul@fokus.fraunhofer.de]
Sent: Monday, November 01, 2004 12:32 PM
To: Bastian Schern
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] UDP Fragmentation Problem
On Oct 30, 2004 at 01:28, Bastian Schern <ml01(a)in-bln.de> wrote:
> Hi everybody,
>
> I've got no success to get a friend in Bogota (Colombia) connected to
my
> SER. He has got a ISDN Internet connection and the UDP packets will be
> fragmented. It seems that the MTU of this connection is round about
400
> to 500 Bytes. Therefore most UDP-SIP packages are fragmented.
> Is SER not able to handle fragmented UDP packages?
The packets are defragmented by the ip stack of the OS, so ser doesn't
really care if they arrive fragmented or not.
> Is it possible to use SIP over TCP with X-Lite?
> Or has somebody another hint for me?
If you plan to use TCP, keep in mind that nat traversal won't work (if
you use it).
Andrei
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We currently do not use an RTP proxy in our service (so the audio does not
ride our internet bandwidth).
Our biggest issue at the moment is the redundancy between two SER servers in
dealing with symmetric NATs (specifically dealing with the individual SER
server unique IP addresses and the far end customer's symmetric NAT).
If we were to use an RTP proxy, as a backup mechanism for dealing with NATs,
would this alleviate the issue of multiple SER servers and symmetric NATs?
Does anyone know a way to send a response to a user agent that will
force it to return to a dial tone. I've been through the RFC's and can't
really find anything and I'm assuming that it's not possible but I
thought I'd ask.
-Evan
Greetings,
I'd like to know a few things. First, can SER control whether a call
is connected or not through radius? For example, SER contacts radius and
radius returns whether the user is allowed to make the call or not, then
SER dials out or not depending on the answer. So this could be per
destination and per user, some users not being able to call to certain
destinations. Is this possible? If so, could someone provide a small
example. Also I'd like to know what acc_rad_request() returns, is it a
bool? What is it's purpose, does it help with the formentioned problem,
or is this acc_rad only to do with CDR style data dumping?
Thank you.
Hello list.
I used the "serctl moni" command and i have found these statistics :
Transaction Statistics
Current: 0 (40 waiting) Total: 584 (0 local)
Replied localy: 1082
Completion status 6xx: 27, 5xx: 7, 4xx: 212, 3xx: 0,2xx: 356
I'm particular interested in the Transactions Statics - Current and the 40
Waiting.
Does this means something?
Thanks in advance.
Ricardo Martinez