dear list,
can any body just give briefly about
what is the difference between the
three basically
iam using forward to redirect the call
so what about the remaining two
thanks for sparing time
________________________________________________________________________
Yahoo! India Matrimony: Find your life partner online
Go to: http://yahoo.shaadi.com/india-matrimony
Hi,
My last two messages that I posted to serusers(a)lists.iptel.org did not appear on
the list. I am wondering if you got my message from about 5 minutes ago.
Please confirm.
Thanks
MS
>From: serusers-bounces(a)lists.iptel.org
>To: masoudsafi(a)hotmail.com
>Subject: Your message to Serusers awaits moderator approval
>Date: Wed, 28 Jul 2004 00:21:24 +0200
>
>Your mail to 'Serusers' with the subject
>
> Re: [Serusers] /var/run/.nscd_socket not found
>
>Is being held until the list moderator can review it for approval.
>
>The reason it is being held:
>
> Post by non-member to a members-only list
>
>Either the message will get posted to the list, or you will receive
>notification of the moderator's decision. If you would like to cancel
>this posting, please visit the following URL:
>
>
>http://lists.iptel.org/mailman/confirm/serusers/68c4426df31b2a7620867772170…
>
_________________________________________________________________
Is your PC infected? Get a FREE online computer virus scan from McAfee®
Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963
Greetings,
One of my user names keep failing on registration, and on the syslog this is
what I am getting. Any ideas where to look for the problem?
Thanks
MS
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: ERROR: parse_uri: bad
port in uri (error at char in state 3) parsed: <sip:1:xxx.xxx.xxx.xxx>(18)
/<sip:1:xxx.xxx.xxx.xxx> (18)
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: get_username(): Error
while parsing R-URI
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: insert_RR(): Error while
extracting username
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: record_route(): Error
while inserting Record-Route line
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: ERROR: parse_uri: bad
port in uri (error at char in state 3) parsed: <sip:1:xxx.xxx.xxx.xxx>(18)
/<sip:1:xxx.xxx.xxx.xxx> (18)
_________________________________________________________________
Is your PC infected? Get a FREE online computer virus scan from McAfee®
Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963
Hi,
I'd like my SER to have the logic to send all (e.g) 474xxxx calls
(555xxxx or whateverxxxx) at any given Asterisk box.
Ideally I'd like to provision (or users self provision) this using the
SerWeb (with some hacking) - rather than changing ser.cfg everytime I
add a * pabx....
Using Aliases (with wildcards???) is one approach. Using R-URI would
probably mean ser.cfg updates.
Are there any other feasible approaches to use?
cheers
Jamie
--
Hi there,
which steps I have to go to change the default MySQL password "heslo" to
"xyz"?
1. Change Password for "ser" and "serro" in DB mysql:
# mysql -u root -p mysql
mysql> update user set password=PASSWORD('xyz') where User='ser';
mysql> update user set password=PASSWORD('xyz') where User='serro';
2. Change ser.cfg:
modparam("usrloc", "db_url", "sql://ser:xyz@localhost/ser")
modparam("usrloc", "db_mode", 0)
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "db_url", "sql://ser:xyz@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
Are this all steps?
I don't think so, because it will not work for me. :-(
Regards
Bastian
Can you make sure you call end_media_session(); before loose_route()?
>>>>>>>>>>>>
Hello List.
My test settings are as follows -
[UA-1]-------[ser + mediaproxy 1]--------------[ser + mediaproxy
2]-----------------[UA-2]
I have the following setup in both the ser.cfg files (ser 1 and ser 2) -
-----------------------------------------
if (method == "BYE" || method == "CANCEL") {
end_media_session();
}
-----------------------------------------
However it appears that when UA2 ends the call, SER 2 calls
end_media_session() - but SER 1 doesn't. When UA1 ends the call, the
opposite happens. Shouldn't both the SERs be calling the
end_media_session()?
Thanks for any feedback.
Regards,
Dhiraj
Hello.
I have a question about the Mediaproxy and NAT'd clients. I am
trying to use the Mediaproxy module to solve the NAT'd clients problems,
until now i have all the scenarios working ok, but i have a question. I'm
using the ser.cfg configuration example in the /modules/mediaproxy/config
directory and i have three clients, one is behind a NAT and the other two
are using public IP's. When i call from the NAT'd client to the Public-IP
client the call is going through the mediaproxy, and that's ok. In the same
way when i call from a Public-IP client to a NAT'd client the call is going
through the mediaproxy, but when i call from one Public-IP client to the
another Public-IP client the call is going through the mediaproxy too.
I think this is not normal. Does the ser.cfg file example work in that
way?
I suppose that in a normal enviroment a call between two Public-IP clients
the media goes directly between them, and there is no mediaproxy in the
middle. Can someone provide me a ser.cfg example that works on that way?.
I'm a liittle bit stuck with this issue for a while.
Thanks in advance.
Best Regards
Ricardo Martinez
Hi,
can anyone help me with this Problem:
I am running a Vega 50BRI Gateway with ser 0.8.14.
Outgoing Calls are handled correctly.
But when I try to make a call from PSTN to the Gateway, I get this
strange error:
---------------------------------------------
SIP Messages:
Vega50----INVITE---->SER:
SER----INVITE---->UAC
Vega50<----TRYING----SER
Vega50----CANCEL---->SER
----------------------------------------------
The Vega 50 Event log show this:
LOG: 01/01/1999 00:05:09.852 ISDN (I)R01C01 incoming
call ref=[f1000020] srce=TEL:0031523338540,DISP:0031523338540 AMT,NAMEC:0031523338540 AMT [0]
LOG: 01/01/1999 00:05:19.857 ROUTER (I)R0bC00 FINDROUTE profile:2(ISDN_TO_LAN) plan:1
call ref=[f1000020] <-- ISDN [1,1] dest=TEL:1004
--> SIP [1,1] dest=TEL:Oliver
LOG: 01/01/1999 00:05:19.967 ISDN (I)R04C01 disconnect 81
call ref=[f1000020]
LOG: 01/01/1999 00:05:19.972 SIP (I)R04C11 disconnect(disc req) 81
call ref=[f1000020]
---------------------------------------------
My Vega 50 Dial Plan for incoming calls:
[planner.profile.2]
enable=1
name=ISDN_TO_LAN
[planner.profile.2.plan.1]
cost=0
dest=IF:99,TEL:Oliver
group=0
name=From_ISDN
srce=IF:01,TEL:<.*>
-----------------------------------------------
The result is that, after the PSTN caller gets disconnected after ca. 8
seconds, the SIP-UAC starts ringing.
What did I wrong ??????
Thanks, Oliver
--
Oliver Zilken
Fraunhofer - FOKUS Research Institute for Open Communication Systems
Competence Centre for Advanced Satellite Communication
Schloss Birlinghoven, 53754 Sankt Augustin, Germany
Phone: +49-2241-14-2947
Fax: +49-2241-14-1005
E-mail: mailto:Oliver.Zilken@fokus.fhg.de
SIP: sip:Oliver@sip.lab.vsat.de
Web: http://www.fokus.fraunhofer.de/satcom