Hi folks!
I have setup my ser.cfg to request www_authentication on INVITE
messages, well, I'm not sure if this is the best solution for allow
ONLY registered users to make calls on my proxy.
Does anybody knows the right way to do this configuration?
Regards
Hello,
I was wondering if it is necessary for a SIP packet from a specific call to
always go through the same server?
For instance, if you have a load balancer distributing requests over a few
servers, it is possible that an INVITE ends up on 1 server while the
following INVITE with the credentials ends up on another, would this be a
problem (ie, break the authorization) or should you use a SIP aware
loadbalancer for this (who looks at the callid for example)?
Assuming the ser servers are setup to use the same userdatabase (and
t_replicate to eachother) the picture would be something like this:
|
--------------
|loadbalancer|
--------------
|
|
--------------------
| | |
------- ------- -------
| | | | | |
| ser1| | ser2| | ser3|
| | | | | |
------- ------- -------
If you setup the servers with the same IP as the load balancer and stop them
from replying to ARP requests for that IP, replying back thru a NAT should
not be a problem.
Just thinking out loud, I could use SER for the load balancing and t_relay
the packets, however that would require some tampering with the VIA records
(and I should use a reply to via in that case to the original IP the SIP
request came from, eg not the load balancer) this way outgoing SIP traffic
would not have to go thru the ser loadbalancer again to get out, hmm, it
might even be possible to use a route-record header to get the packets back
at the correct server...
Kind regards,
E. Versaevel
Hi,
Here is my problem in user subscription using serweb:
1st phenomenon:
I use User Management Page to add a user by means of Subscription. After
I have added a user with username "thomas", and after receiving the
confirmation email and clicking on the provided link, the error message
occurs:
"400 Table 'aliases' not found in memory, use save("aliases") or
lookup("aliases") in the configuration script first
We regret but your officehk123.vttsys.com confirmation attempt failed.
Please contact info(a)iptel.org for further assistance."
However, I can login the user management page using "thomas".
Question:
- What is the cause of such error?
- Why could I login the User Management Page?
2nd phenomenon:
I go to the AdminLogin page to delete the user "thomas". The user
account is removed from the table.
However, strange thing happens.
When I go to the User Management Subscription page to add a user
"thomas", error message displays:
Sorry, the user name 'thomas' has already been chosen. Try again.
Question:
- user "thomas" is supposed to be deleted. However, I am not allowed to
add it again. Why does this happen?
Anyone can help me?
Thank you very much.
Thomas
Hi,
I encounter the following error message after clicking the confirmation
link received through email after user subscription:
officehk123.vttsys.com User Management
400 Table 'aliases' not found in memory, use save("aliases") or
lookup("aliases") in the configuration script first
We regret but your officehk123.vttsys.com confirmation attempt failed.
Please contact info(a)iptel.org for further assistance.
Thomas
Hi,
After chown 666 /tmp/ser_fifo, I can solve this problem. Thanks.
Thomas
On Thu, 2004-12-02 at 09:31, Richard Machida wrote:
> I believe you need worl rw perms on the fifo file (/tmp/ser_fifo). It
> needs to be reset everytime SER is restarted.
> Hope this helps....
>
> Richard
>
>
> On Wed, 01 Dec 2004 16:19:56 +0800, support <support(a)cybertel.biz> wrote:
> > Hi,
> >
> > I have just installed serweb successfully on FC 2. Initially, I notice
> > there is one user called admin show on the index page, but when I
> > clicked into account, an error shows in red color:
> >
> > sorry -- cannot open write fifo
> >
> > I try to add a user, and the same error message occurs.
> >
> > What is the problem?
> >
> > Thomas
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
hi,
How can i configure my sip proxy to automatically know that the PSTN gateway
is down and transfer all calls to another gateway?
thank you
regards,
ron
I'm curious what brand load balancer you would use, would it be IP
based. We tried out a Cisco SLB and had no luck, mainly because it would
NAT to the servers (more trouble than it's worth?). We were thinking of
using a heartbeat type failover, similar to what you would do for MySQL:
http://linux-ha.org/download/
Has anyone tried this method? We're more concerned about the high
availability than anything.
-----Original Message-----
From: E. Versaevel [mailto:erik@infopact.nl]
Sent: Wednesday, December 01, 2004 7:24 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Loadbalancing / high availability
Hello,
I was wondering if it is necessary for a SIP packet from a specific call
to always go through the same server?
For instance, if you have a load balancer distributing requests over a
few servers, it is possible that an INVITE ends up on 1 server while the
following INVITE with the credentials ends up on another, would this be
a problem (ie, break the authorization) or should you use a SIP aware
loadbalancer for this (who looks at the callid for example)? Assuming
the ser servers are setup to use the same userdatabase (and t_replicate
to eachother) the picture would be something like this:
|
--------------
|loadbalancer|
--------------
|
|
--------------------
| | |
------- ------- -------
| | | | | |
| ser1| | ser2| | ser3|
| | | | | |
------- ------- -------
If you setup the servers with the same IP as the load balancer and stop
them from replying to ARP requests for that IP, replying back thru a NAT
should not be a problem.
Just thinking out loud, I could use SER for the load balancing and
t_relay the packets, however that would require some tampering with the
VIA records (and I should use a reply to via in that case to the
original IP the SIP request came from, eg not the load balancer) this
way outgoing SIP traffic would not have to go thru the ser loadbalancer
again to get out, hmm, it might even be possible to use a route-record
header to get the packets back at the correct server...
Kind regards,
E. Versaevel
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
when I login to my page http://xx.yy.zz.ww/admin/index.php
and I insert:
username = admin
password = heslo
I receive: Bad username or password
I have created mysql tables with ser_mysql.sh,changed the file config.php
and I have check with the tool mysql_administrator and mysql_query_browser all the tables.
the user admin exist with this data:
username=admin
password=heslo
...
perms=admin
...
ser and serweb run in the same machine,
my hostname it's same of ser and sip domain
and finaly I'm using this version of serweb:
serweb_2004-07-27.tar.gz
Can You help me, Please?
Thank You
--
___________________________________________________________
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Hi,
I'm setting up a ser + * + cisco 2600 architecture where SER's clients got
a 72.. (4 digits) plan number and Cisco's users got a 1... (4digits).
I got 2 x-lite clients, MSN Messenger and a SJPhone registered to SER; A
analog phone, an IP Phone and a Wireless IP Phone registered to my cisco
gateway.
Below are my troubles:
1 - Calls between 2 SER's clients are OK; caller got a tone ringing but
nothing for the called (no indication and no callerID). Do I need to add a
setcallerdId in my extensions.conf ? Previously, I was not obliged and I
had callerid and an indication;
2 - When a x-lite client calls the analog phone linked to cisco's gateway,
it's OK (ringing, answering, hanging up). life is great !!!!;
3 - When a x-lite client calls an IP Phone (wired or wireless), called got
the ringing, caller heared the ringing tone but when caller tried to
answer, call is destroyed;
4 - When an IP phone (wired or wireless) send a call to an x-lite client,
it is not ringing in the called side (no ringing tone, no callerid
indication) and asterisk got a such message:
== Spawn extension (default, 7201, 1) exited non-zero on
'Local/7201@default-9565,2'
and then destroyed the call
Destroying call '502DA4E9-42E211D9-8AACFF3B-E217174C(a)10.0.0.8'.
I thaught, I got some problems in my extensions.conf may be I'm wrong !!!
my configurations files are linked.
Thanks in advance
Ahmed Boreau
D. Informatique
ESMT
