What setting are required for making calls using Audiocodes MP-108 FXS with
Ser.
I am able to register the Audiocode device with Ser but Ser rejects all my
call attempts.
If I put Username as ENDPOINT Number I am able to use that particular port
for making calls but what about others?
Do I have to register aliases in SER for each port of audiocode?
Thanxs
Deepak Singhal
Hi,
Just saw an article on
http://www.zdnet.de/news/tkomm/0,39023151,39132110,00.htm?h (in German)
where they say that AVM and Iptelorg developed some ISDN features in SIP
like "callback on busy", "call waiting", "cold hold", "deflection" and
"3way conference".
I was just wondering:
- how the first one (callback on busy) is implemented and if it is
possible to do this with the standard ser and without a b2bua
- why the other features are big news since they are possible with some
help of the involved UACs anyway...
Andy
hello
i am not getting accounting stop on Radius. only
accounting start is comming. can any one tell me what
is the problem in my configuration.
thanks in advance
Kamran
ser.cfg
-------------------------------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
port=5060
fifo="/tmp/ser_fifo"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_radius","service_type",15)
modparam("acc", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "log_level", 1)
modparam("acc", "log_flag" , 1)
modparam("acc", "radius_flag", 1)
#end of radius configuration
route {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
setflag(1);
if (method=="INVITE") record_route();
if (method=="INVITE") {
rewritehostport("mygwIP:5060");
}
if (uri==myself) {
if (method=="REGISTER") {
if(!radius_www_authorize("")) {
www_challenge("","0");
break;
}
save("location");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
__________________________________
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- Make sure you are not behind a Symmetric NAT. If so, you're dead. STUN
does not work with Symmetric NAT.
If a UA is behind Symmetric NAT, and
UA use STUN, and
SER have [RTP/Media]Proxy to handle Symmetric NAT,
this UA should be fine, right?
Thanks,
Mohammad
--------------------------------------------------------------------
mail2web - Check your email from the web at
http://mail2web.com/ .
Can I have two server, each having one network interface with one ip?
Thanks,
Mohammad
Original Message:
-----------------
From: Lucas Aimaretto lucas(a)cyneric.com
Date: Thu, 7 Apr 2005 15:56:13 -0300
To: info(a)beeplove.com, serusers(a)iptel.org
Subject: RE: [Serusers] STUN server
> Do I need to have two interface on the server?
> Or I can have two different server.
You need to have two IP address in the server. Whether you use two
network cards with an IP on each, or two IPs address on just one network
interface, it is up to you. I'm using two different Ips in just one NIC.
It works fine.
Regards,
Lucas
--
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Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 06/04/2005
--------------------------------------------------------------------
mail2web - Check your email from the web at
http://mail2web.com/ .
Hello,
I have a problem with the gateway cisco 1760. I make a capture and the
arrive to the gateway but this say me "BAD REQUEST -IP invalid" and I
don't what to do
My configuration is:
Using 1454 out of 29688 bytes
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname ToIP
!
!
username GW password 0 GW
ip subnet-zero
!
!
ip name-server 172.17.1.20
!
!
isdn switch-type basic-net3
isdn voice-call-failure 0
!
voice call carrier capacity active
!
voice class codec 1
!
ip name-server 172.17.1.20
!
!
isdn switch-type basic-net3
isdn voice-call-failure 0
!
voice call carrier capacity active
!
voice class codec 1
codec preference 1 g723r63
codec preference 2 g723ar63
codec preference 3 g729br8
codec preference 4 g729r8
codec preference 5 g711alaw
codec preference 6 g711ulaw
!
!
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
description Line
ip address 172.17.1.2 255.255.255.0
speed auto
!
!
!
!
!
interface FastEthernet0/0
description Line
ip address 172.17.1.2 255.255.255.0
speed auto
!
interface BRI0/0
description RDSI 932531082
no ip address
isdn switch-type basic-net3
!
interface BRI0/1
description RDSI 932531081
no ip address
shutdown
isdn switch-type basic-net3
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.17.1.1
no ip http server
ip pim bidir-enable
!
!
!
call rsvp-sync
!
voice-port 0/0
compand-type a-law
cptone ES
no ip http server
ip pim bidir-enable
!
!
!
call rsvp-sync
!
voice-port 0/0
compand-type a-law
cptone ES
!
voice-port 0/1
compand-type a-law
cptone ES
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
description 932531081
destination-pattern .........
no digit-strip
port 0/0
!
dial-peer voice 200 voip
session target sip-server
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
!
dial-peer voice 200 voip
session target sip-server
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:172.17.1.20:5060
!
!
line con 0
line aux 0
line vty 0 4
exec-timeout 0 0
login
!
ntp clock-period 17179997
ntp server 130.206.42.224
end
Thanks.
--
Miquel
Hello,
I'm a new user and I have a question. How can I control accounts, I use Serweb, but in admin when i press the "accounting" it doesn't work. And anybody can make calls using sipphone without loggin in (not registered in mysql but it can make a call), how?
Thanks...
Sorry if this is off-topic, but I know there's a quite a few smart
people who frequent these groups, and I was thinking that it'd be a good
place to ask.
We have an opening for an experienced PERL programmer. If you (or anyone
you know) is interested, please feel free to email me for more details.
--
==========================================
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600 Fax: +613 99401650
==========================================