oki wont in some way to make a call from sems and make 'connection' between person X(SIP softphone) and IVR script
i found one bash script in SER /examples/ctd.sh i try to conect person
sip:menta@xxx.xxx.xxx.xxx with sip:ivr@xxx.xxx.xxx.xxx
in first instance of the ser server i have rule that say if user is not online forword this request to second instance of the ser
in second instance of ser i have the rule that say :
....
else if( uri =~ "ivr" ){
log(1, "Time to process IVR script\n");
if(!vm("/tmp/am_fifo","ivr")){
log("could not contact IVR engine\n");
t_reply("500","could not contact IVR Engine");
};
}
...
that must forword to SEMS server..
In this way everything works perfect if some one call ivr(a)xxx.xxx.xxx.xxx
but when i try to initiate the call with ctd.sh i gave got this error:
2(1798) ERROR: parse_uri: bad uri, state 0 parsed: <Via:> (4) / <Via: SIP/2.0/UDP 80.72.85.178;branch=z9hG4bKb921.b0ba3c52.0> (59)
2(1798) ERROR: fifo_uac_error: ruri invalid
2(1798) ERROR: fifo_server: command must begin with :: To: <sip:menta@80.72.85.178>;tag=6709f4d5
2(1798) ERROR: fifo_server: command must begin with :: Call-ID: 10942351191984.fifouacctd
2(1798) ERROR: fifo_server: command must begin with :: CSeq: 2 REFER
2(1798) ERROR: fifo_server: command must begin with :: Refer-To: sip:ivr@80.72.85.178
2(1798) ERROR: fifo_server: command must have at least 3 chars
2(1798) ERROR: parse_uri: bad uri, state 0 parsed: <Via:> (4) / <Via: SIP/2.0/UDP 80.72.85.178;branch=z9hG4bKb921.b0ba3c52.0> (59)
2(1798) ERROR: fifo_uac_error: ruri invalid
2(1798) ERROR: fifo_server: command must begin with :: To: <sip:menta@80.72.85.178>;tag=6709f4d5
2(1798) ERROR: fifo_server: command must begin with :: Call-ID: 10942351191984.fifouacctd
2(1798) ERROR: fifo_server: command must begin with :: CSeq: 3 BYE
2(1798) ERROR: fifo_server: command must have at least 3 chars
Phone at menta(a)80.72.85.178 ringing when he answer server throw this error.
and when i use this bash script all the configuration in ser was scipet i mean its dont vorword the call to second instance at all ...
and to be honost i dont have any fogest idea why that happend
What can be wrong ?
-----------------------------------------------------------------
http://www.lex.gbg.bg - За да знаете правата си!
I have Mysql with ser 0.8.12 installed.The accounting works but I have a
problem: It seems that not all the accounting information appears in acc
in mysql..For example: If I place a call and I go to mysql in acc I don't
see that call accounted...What is the problem ?
Sirs:
For Billing, is BYE related to the call ending time in the SER
accounting tables? If it is not, how can BYE time be obtained?
Thanks
Juan G. Castañeda
I am still chasing a problem with using the CC-Diversion header to
redirect a call
back to my service provider to an Octel system hanging off their switch.
I am a Verizon Centrex customer. Inbound calls are presented to my
Cisco 2620XM
router via a PRI . The called party number has 5 digits. When using the
SER and
CC-Diversion header to redirect an unanswered call to our Octel
voicemail system
I repeatedly get the following:
"Cause i = 0x809C - Invalid number format (incomplete number)"
After some digging I suspect that Verizon wants me to send them more
digits so
they can forward the call to the correct mailbox. This is consistent
with the above
error. The problem is no one, did I say no one, seems to know how many
digits
they want.
So the question is this. How can I manipulate the "Original Called
Number" value
in the outbound ISDN setup message to contain more or less digits? It
seems like this
value is extracted from the value in the To header. Is this correct? In
the ISDN
messages the original called number is always the 5 digit called party
number.
Thanks,Steve
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-7903
fax: 215-898-9348
sip:blairs@upenn.edu
Hello,
I have a question concerning the nathelper module and rtpproxy.
Here is the network layout :
Sipphone1 <----> Firewall with NAT <------> SER with nathelper <---> Sipphone2
(public IP) public IP/private IP + rtpproxy private IP
(private IP)
I understand that the regular way of using nathelper/rtpproxy is to
put them on a dual interface computer (one internal interface and one
external one). In my case it is not possible, the SER box is inside
and has only one internal interface.
I am puzzled at how I should configure SER and rtpproxy to allow
sip calls from outside to inside.
Currently the sip trafic is fine, ie I call from outside to inside,
and my internal phone is ringing. The problem is with the rtp trafic.
I can't find the way to make nathelper/rtpproxy change the rtp IP to be
used for the call.
I am using CVS version of ser/nathelper/rtpproxy. Whatever the config
I try (examples, and mails from this list) the IP in the sip packet
stays the one from Sipphone2 (where it should me the external
interface ip from the firewall).
Another way to formulate my problem is : how could I tell nathelper
which IP it has to use to replace Sipphone1's ip for rtp trafic ?
Thanks !
Guillaume
Hello,
I have a problem, with INCOMMING call :
UA1 >> SRV_IPTEL >> GATEWAY CISCO AS5300 >> PSTN : IS GOOD
PSTN >> GATEWAY CISCO AS5300 >> SRV_SIP >> UA1 is not connected
If UA1 is disconnected, when I try to phone to UA1, I have a lot of error in
CDR on my GATEWAY.
What can I write in ser.cfg to says, when a UA1 is not in LOCATION send an
ERROR at the GATEWAY, and THE GATEWAY send a error on the PSTN.
Thanks in advance
Nicolas RUIZ
VIVACTION
France, PARIS
I want to cofig ser as a outbound server to resolve all nat problem. The follow picture is the system architecture.
uac--nat-->outboud1(ser1+nathelper+rtpproxy)------------->ser(main)
uac--nat-->outboud2(ser2+nathelper+rtpproxy)-------------|
uac--nat-->outboud3(ser3+nathelper+rtpproxy)-------------|
Can someone know if ser can rewrite the uac's contact as itself ip address and
forward to main ser? Nathelper can rewrite sdp ip/port to it own ip/port. But it
only rewrite contac with nat's ip and port.
Jiangzhou
Hello,
I am trying to make media proxy to direct all media packets through my
ser.I have a computer with ser which does all the routing sequences and I
have another computer with ser and with the mediaproxy installed .The
problem is that i don\t know how to configure one of those 2 sers or just
one of them with mediaproxy.I have the ser.cfg from the media proxy module
but I am not able to find any further documentation regarding ser
configuration with media proxy.
