I can't get media proxy running.I tried everything in my ser.cfg ( I
looked over the ser.cfg example included in the mediaproxy module ) but
still doesn't work.I mention that the test I've made were on my cisco as
5350 gateway.I looked with show call active voice brief while I was
speaking from a sip phone through my ser server and the ip on that active
voice brief entry wasn't the one of my ser server's as it was supposed to
be..Can anyone help me or give me two-three lines of a working config ?
Thanks
HI ALL;
I have couple of sip UA registered in my ser box which is enabled with "acc module". I just want to do simple acconting when they call each other
and to keep track the duration of call
can anybody give me a sample ser.cfg which do accounting for just registered users
Warmest Regards
mohammad
HI ALL;
I have couple of sip UA registered in my ser box which is enabled with "acc module". I just want to do simple acconting when they call each other
and to keep track the duration of call
can anybody give me a sample ser.cfg which do accounting for just registered users
Warmest Regards
mohammad
HI ALL;
i have been trying to load ser with acc module for radius accounting but it seems my dictionary is not complete
ser says:
0(21087) ERROR: acc: can't get code for the Failed attribute value
0(21087) init_mod(): Error while initializing module acc
Appreciate Any Suggestion
Regards
mohammad
ok i try the bot plugin but have strange error in SDP parsing or maybe in
packet constructing
i am using
version: ser 0.8.99-dev6 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.193 2004/09/19 20:22:51 andrei Exp $
main.c compiled on 18:00:46 Sep 22 2004 with gcc 3.2
and SEMS from CVS
OK this is the error:
(6997) DEBUG: write_to_fifo (SerClient.cpp:244): Write to fifo: completed
(6997) DEBUG: getReturnCode (AmRequest.cpp:357): response from Ser: 200 OK
(6997) DEBUG: getReturnCode (AmRequest.cpp:391): headers : <Via: SIP/2.0/UDP
83.148.127.2;branch=z9hG4bK72e6.bd7d9c36.0
To: sip:menta@83.148.127.2;tag=99327ab6eb4c53f7a991c11e10f2f90f.8f07
From: sip:bot@83.148.127.2;tag=00001B1919BCF4D7
CSeq: 11 INVITE
Call-ID: 00001B1945BBEB97(a)83.148.127.2
Contact: <sip:menta@83.97.31.127:5061>;expires=2582,
<sip:bot@83.148.127.2:5060>;expires=3600
Server: Sip EXpress router (0.8.99-dev6 (i386/linux))
Content-Length: 0
Warning: 392 83.148.127.2:5060 "Noisy feedback tells: pid=6987
req_src_ip=83.148.127.2 req_src_port=5060 in_uri=sip:menta@83.148.127.2
out_uri=sip:menta@83.148.127.2 via_cnt==1"
>
(6997) DEBUG: getReturnCode (AmRequest.cpp:397): found 'To:
sip:menta@83.148.127.2;tag=99327ab6eb4c53f7a991c11e10f2f90f.8f07'
(6997) DEBUG: getReturnCode (AmRequest.cpp:410): found To-tag:
'99327ab6eb4c53f7a991c11e10f2f90f.8f07'
(6998) ERROR: parse_sdp_line_ex (AmSdp.cpp:415): parse_sdp_line : parameter
'v=' was not found
(6998) ERROR: run (AmSession.cpp:183): 400 session description parsing
failed
for the first try i use eStara UAC after that i put SJphone on SJP i got
user not found but mybe this is becouse dont know why but SJphone dont send
register packet
after that i put a X-Pro and got the same error like eStara
any suggestion what is actualy the problem and how i can fix it ?
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HI ALL;
Can anybody explain what exactly is "statefull proxy"???????
I read the ser related documents about that topic but I have not understood very well
Warmest Regards
mohammad
Hi, all
If I want to let my ser call to telephone by PSTN gateway.
What kinds of PSTN hardware require??
sorry, I am newly to contact PSTN.
So, if I have a modem card ,could my ser can call to telephone?
Thanks.
For some reason SER won't load on my 64 bit Fedora Core 2 machines but it
will work fine on all of my other 32bit Fedora Core 2 machines. Anyone know
if this a known problem with SER? I haven't been able to find any info on
it so I'm guessing I'm just doing something wrong. Any help would be
appreciated.
WARNING: no fork mode
0(9861) DEBUG: init_mod: sl_module
stateless - initializing
0(9861) DEBUG: register_fifo_cmd: new command (sl_stats) registered
0(9861) DEBUG: MD5 calculated: 88ea29e41d7a330a3a8d2ddf294ed2ce
0(9861) DEBUG: init_mod: tm
0(9861) TM - initializing...
0(9861) Call-ID initialization: '060d4f9b1d214389'
0(9861) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered
0(9861) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered
0(9861) DEBUG: register_fifo_cmd: new command (t_hash) registered
0(9861) DEBUG: lock_initialize: lock initialization started
0(9861) Segmentation fault
Thanks,
Mike
Hello experts,
I am wondering if there is a possibility to run MySTUN (STUN server) on a
Linux platform using only one network card.
As far as I understand, the two server addresses require two network cards
and a router to combine the traffic from two network cards towards the
clients.
But maybe there is some trick from some networking expert ...
Any suggestions?
Regards
Franz
Is it possible to strip the '+1' from the FROM and TO fields? strip(2) only
removes it from the INVITE. I need to forward to a gateway that doesn't
support e.164.
