Does anyone have a config for an Aastra 9133i IP phone to a SER box? I am
trying to configure one. I can make calls to the Aastra but can't make calls
from the Aastra. It registers properly.
Thanks
Bill
Hello,
I have been asked many times where is the best place to obtain
support and ask questions for CDRTool. As CDTool is at this moment
primarily designed and tested for OpenSER deployments I have decided
that OpenSER users mailing is the best place to make announcements
for new releases. There is of course no guaranteed support, as you
may expect for a free product, support requests will however not go
unheard.
Unless there is no objection or complaint, I will keep this mailing
list updated about CDRTool new releases from now on.
Here it is:
CDRTool 5.0.5 has been released. The software can be downloaded from:
http://download.dns-hosting.info/CDRTool/
Changelog:
* Fixed determination of previous year in
update_raddact_record_mediaproxy
Consolidate SET statements to minimize binary log entries. Fixed
bug in
update_raddact_record_mediaproxy, the record values were reset
after
execution of update on the first table causing failure to update
the
previous table. setup/radius/OpenSER/radius_accounting.proc must be
reloaded into the MySQL server:
mysql -u root -p -h sipdb radius < setup/radius/OpenSER/
radius_accounting.proc
* Use same CDR structure for MaxSessionTime and DebitBalance
* Change logging for prepaid actions to fit one line
* Normalize tries previous month if no record has been updated
* Mention how to see prepaid account status
* Updated rating docs
* Moved E164 class to phplib/local.inc
* Show accounts with quota that exceeded a certain treshold
scripts/SER/quotaShowAccounts.php treshhold
* Added documentation for the Quota system (doc/QuotaSystem.txt)
* Use normalization lock per table using GET_LOCK() server function
http://dev.mysql.com/doc/refman/5.0/en/miscellaneous-functions.html
This provided faster web response when multiple clients access the
interface
* Fixed confirmation for delete operations in rating tables when
global
vars are turned off in php.ini
Regards,
Adrian
Hi,
yesterday i've began to upgrade openser from 1.0.1 to 1.1.0. Now i got a
strange situation when registering with two different clients with the
same sip account.
in the OK Message from the sip proxy i have a strange contact field:
Contact: <sip:012345@xxx:5070>;q=0.9;expires=3600,
<sip:012345@xxxx:63036>;expires=25
i think in the OK Message should only be one contact adress?
Hi all,
Merry Xmas !
I'm just a small question if some people are still working :-)
I try to make a load balancing solution with SER and Asterisk.
I use the dispatcher module and I set on asterisks servers the default
gw as SER.
All requests go through the SER to the asterisks and on the SER I'm have
a line like
If source-ip = Asterisk source IP do ...
And I want to do something like forward to the TO FIELD ...
Do you know how can I do this ?
Thanks a lot for your help,
Thomas
Hello,
I have a few questions about how to secure SIP messages...
IPSec or TLS can be used to guarantee privacy, integrity and
authenticity of transmitted data. But SIP is only payload/data for
these protocols.
Digest Authentication does only guarantee authenticity...
S/MIME seems to be interesting but I don't know any implementation in
a SIP client.
My questions:
- Is there any mechanism that prevents me from corrupting or faking
SIP messages?
- Is it possible to create a kind of binding between the certificates
used for TLS/IPSec and the SIP accounts?
Thanks for your answers...
Best regards,
Steffen
Hello,
Does anyone have recommendations for a SIP based termination provider in
Europe and/or Australia? I am looking specifically for a wholesale, carrier
grade provider. I have found several which appear to be asterisk based and
quite small. What I am looking for is more of a traditional carrier that is
SIP enabled as opposed to a smallish mom and pop shop.
Thanks in advance.
T.R.
Hello all,
I'm trying to use a SER server for the following:
|
| | |
ISP #1 ISP #2 Private Link
Private LINK
NATed SIP SIP Clients 172.18.xxx.yyy
172.16.5.xx
Clients |
| |
|
| | |
| |
| |
----------------------------------------------------------------------------------------------------
|
----------------------------------------------------------------
SER with MEDIAPROXY
----------------------------------------------------------------
|
|
|
================
Asterisk (172.16.5.1)
=================
|
|
PSTN
I need asterisk to handle all the calls to the PSTN and event the SIP
calls between them. I need to use the SER server as a tool to receive
calls from any of the environments that I draw on top, and all
communicating with Asterisk, even I also have a backup Asterisk box,
that I can use in load balance with SER.
I saw that there is a function "rewritehostport()" which I'm pretty sure
it won't work since I will not keep the call in the SER server. I have
made the NAT configuration of the Getting Started Guide, but I now that
this is just part of what I need to build the complete solution.
Can anyone please give me a hint in what is that I need to acomplish
such task
Thanks in advance
Gerardo Amaya
Hi, all
I install mysql server and SER in my FC6
when I run SER with mysql to do authenticate
#ser -f auth-mysql.cfg (the SER getting started example)
it prompt me that can't connect to local MySQL server through socket
'tmp/mysql.sock'
so I modified the /etc/my.cnf' file to change socket=/tmp/mysql.sock
but when i start SER and use serctl to add a user
ERROR 2002 (HY000): Can't connect to local MySQL server through socke
'var/lib/mysql/mysql.sock'
was prompted
I don't know if I could open two sock at the same time or how can I let SER
or serctr
chage the path to fit another.
Did someone can show me some clue or solution?
thanks!!
Can someone help?
I am fairly new to SER and Mediaproxy and can't find a detailed explanation
of how nathelper etc work, so I'm struggling!
My config is:-
SER 0.9.6
Mediaproxy 1.7.2
Fedora Core 4 86x64
SER and Mediaproxy run on the same server which acts as a gateway to a
remote Sonus PSTN gateway over a VPN.
I also use Asterisk on my local network. The SER server acts as a router
between Asterisk servers and the PSTN via the Sonus. This works fine but my
problem is when I attempt to use X-Lite 3.0 UA behind NAT on a remote
network. Asterisk is not used in this config but proves that two way audio
is fine using SER and the Sonus gateway.
Problem: X-Lite behind NAT on remote network to Sonus PSTN gateway via SER
and Mediaproxy, I get only one ay audio. The audio is fine from X-Lite to
PSTN but not in the reverse direction.
Thanks for your help