Hi Daniel-Constantin,
I am Tai Vo. I am trying to run the default SER script as a SIP Proxy.
However, I could not make any Register attemp(s). Could you please let
me know why and how I can run properly the default SER script which
is downloaded from IPTel.
Thank you very much for your help in advance.
Best Regards,
Tai D. Vo
tducvo(a)yahoo.com
Hi Everybody
When I run ser using the ser.cfg default configuration, all the users can connect to my mysql, but When I uncomment the lines who asks for authentication, they can't Authenticate anymore ...
First When they are authenticated, I can see, via serctl that there are users connected, and see their usernames, thats to say, mysql is running well but I can't make my users auth at it... Is there a way to test the sip authetnication or, to test the mysql auth without ser ??
Thanks in Advance
PS: My ser.cfg is like this (Attached File)
Hi All,
Just a quickie, I'm having MEGA problems getting SER/SEMS operating as a
voicemail server. I'm running SER as a proxy on 5060 and SER/SEMS on 5090.
I can get the calls to divert to SER/SEMS on 5090 if the subscriber is
unavailable, but I'm having huge problems getting SER/SEMS running properly.
The example conf's in the SER-SEMS howto dont seem to work on my set up, and I
keep getting database errors thrown up.
I have mailed the SEMS list, but I wondered if anyone here has a working
SER/SEMS Voicemail configuration, would you be kind enough to post your
ser.cfg and sems.conf so I can compare to try to see where I'm messing it up?
Many thanks,
Ian
Hello.
I'm running ser 0.8.14 with serweb on debian.
A user can subscribe, but when confirming to the email sent out, I get this
error:
"400 Table 'aliases' not found in memory, use save("aliases") or
lookup("aliases") in the configuration script first ".
How to go about it?
Amos.
I am trying to manipulate the 'from' header to add a leading zero - our
PSTN gateway does not send it over in the from header for CLI.
If I use the replace function - replace ( "From:.*<sip:", "From:
<sip:0"), then this is OK for the subsequent messages that are
forwarded, but not for the current message that is in memory and this is
the one that is being put into the database for accounting.
Looking at the source it looks like the replace function is effecting
the main buffer, but if the from header has already been read then I can
manipulate it ! !
Am I right in this thinking. Is there a way around it other then writing
my own module ? I'm currently running the ser from the development CVS
Thanks
Simon
Hi all,
Radius Acct now works but i have a problem with "stale calls". Perhaps some
times SIP Client forget to send BYE or CRASH and, in those cases, RADIUS
Daemon don't set MySQL DB "radacct" correctly (end remail 00:00:00 00-00-00).
How i can set a timeout to avoid this problem ?
Oz
--
------
O-Zone ! www.zerozone.it
Hi All,
Is it possible to define the RTP port range that ser
uses? I'm interested in this information in order to
setup a firewall policy.
Any help is highly appreciated.
Regards,
Lakmal
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HI, ALL!
Happy New Year!
BUT ... :-)
After registered with a pc installed SER(public IP 'A'), I could not
get any sound on my own pc behind NAT(public IP 'B'). So, I wanna SER
be Registerar and another pc (public IP 'C') installed siproxd as
OUTBOUND PROXY. (I could hear something when changing
outbound-proxy-setting to "fwdnat.pulver.com:5082").
SO, questions are:
1. Can I use Siproxd as OutBound Proxy in public IP?
2. If so, How Can I setup the config file?
#########################
### siproxd.conf of mine ###
if_inbound = eth0
if_outbound = eth0
sip_listenport = 5082
daemonize = 0
( and some pid, rtp confs ...)
######### end ############
3. According Jan, SER can also be used as outbound proxy, How can
I setup it ?
Thank you!
Any adviced advanced!!!
--
Regards,
Embebo
2005/01/05
Hi,
Happy new year to all!
I run into one case that ser doesn't cancel pending INVITE after CANCEL is
received. When a caller sends INVITE to ser, ser forwards to the contact
location in its record. If a contact is not expired, but the phone is not
available somehow, ser keeps trying to send INVITE. During this time, the
caller gets no response because no one sends 180/183. At this time, the
caller hangs up and sends a CANCEL. Ser sends back "SIP/2.0 200 ok -- no
more pending branches". The problem is that ser still keeps sending INVITE
which should be cancelled by the CANCEL. In some cases, these INVITE
eventually reaches the phone. The callee phone starts ringing even though
the caller has hanged up already.
Is there any fix to the problem?
Thanks,
Richard
