Hi All.
I came across this posting
http://lists.iptel.org/pipermail/serusers/2004-August/010832.html where Bogdan
suggested sending REGISTER messages to multiple ser proxies using something
like this:
append_branch( dst1 );
append_branch( dst2 );
t_replicate( dst3 );
How would I best solve a problem where if I were to have say 10 ser proxies and
wanted to avoid changing the ser.cfg everytime I added/removed a new sip proxy
to the farm? I'm thinking it would be best to do this with a custom module
which would look up rows in a MySQL table. Each row would be the location of a
ser proxy which would be used in a call to append_branch(). This way I could
add new ser proxies to the farm without modifying ser.cfg on every machine and
in my ser.cfg file I could replace the above snippet with something like this:
append_proxy_locations();
Does anyone see pitfalls with this? Is there a better solution?
Regards,
Paul
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Hello:
I just installed SEMS, sems_2004_01_04, on a new SER 0.8.14 stable
system. I have the same version of SEMS running on my other
system, a 0.8.12 stable system, that is working fine.
On the 0.8.14 system I see the call is sent to the defined failure
route but
no log messages appear from the second instance of SER that is suppose to
interface with SEMS.
Is there someplace where how-to use SEMS with this version of SER is
documented? I don't get any error messages in my logs when starting either
SER or SEMS.
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
Hi Guys,
can i use SER in a transparent mode like a squid, with a iproute-redirect?
My user currently using an
asterisk for incoming and outgoint calls, but i need to do an accounting of
every call (call-lenght) on
the firewall, is this possible with SER, including some kind of
"pass-through" authentication/registration
to asterisk? I don't have direct controll over the "backend", so it's not
possible to setup the users on SER...
If someone has already done something like this i would really appreciate a
ser.conf as a template :-)
Regards,
Andreas.
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Hi everybody,
I've got no success to get a friend in Bogota (Colombia) connected to my
SER. He has got a ISDN Internet connection and the UDP packets will be
fragmented. It seems that the MTU of this connection is round about 400
to 500 Bytes. Therefore most UDP-SIP packages are fragmented.
Is SER not able to handle fragmented UDP packages?
Is it possible to use SIP over TCP with X-Lite?
Or has somebody another hint for me?
Regards
Bastian
Hi all.
Is there a new and improved way to tie sems to ser? I can't find any source code in CVS for the
vm.so module so I assume vm.so has been deprecated.
Regards,
Paul
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Hello,
I have Micronet SP/5050s as a gateways to PSTN (connection to PSTN via
PBX), SER as a proxy server and some voip phones. Next picture shows my
topology:
_________ _______ _________ _____
|SIP Phone|___| SER |____|SP 5050/S|____|PBX1 |____ PSTN
|_________| |_______| | |____1____| |_____|
|
| . . .
| _________ _____
|__|SP 5050/S|____|PBXn |____ PSTN
|____n____| |_____|
(in other location)
If voip user calls some number, his UA sends INVITE message (to SER)
with Request-uri=To-uri=number1(a)domena.sk.
My SER overwrite Reguest-uri to number2@ip_address_of_micronet an sends
this INVITE message to Micronet. INVITE includes Reguest-uri which is
different from To-uri. I need dial number2 in PBX.
Note: Number1 may contain prefix - it tell which PBX use. SER must
prepare real PSTN number - number2 and send INVITE message to corect
gateway.
My problem is that micronet gateway dials numbers from To-uri field, not
from Request-uri field of INVITE message. So it dials number1. All
numbers managed by SER are not available.
I can overwrite called number (number1 from To-uri) in micronet gateway
too - gateway uses prefixes, but only 10 - it is few for me. And I need
more micronet gateways with different prefixes (I have more PBX each in
other place) - micronet's prefixes decentralize number logic from one
SER to more gateways.
I was trying to overwrite To-uri field of phone's INVITE message in SER,
but if SER sends other message with overwrite To-uri (200 OK, ACK, reply
from micronet) to phone, phone rejects it - message has wrong To-uri
field.
When I has Cisco 2600 as a PSTN gateway - it dialed numbers from
Reguest-uri field, but Cisco is more expensive.
Do you know some solution for me or other gateway, which can I use -
gateway must dial numbers from Reguest-uri field and its price is circa
300$ (PSTN gateway with 6 FXO ports).
Thank you,
Regards,
Laco
Is there any solution to the problem of no BYE message being recorded by the acc
module when the callee hangs up first ? I have found the question asked several
times in the archives but no answers.
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Dear All:
Would you please check these lines if any bad config.
When I add these lines, I always got two errors. Thanks!
if ( search ( "From:.*sip:0944021[0-9][0-9][0-9]@" ) &&
!( search ( "Call-ID:.*@prepaid.example.com" ) ) &&
( method=="INVITE" || method=="CANCEL" || method=="BYE" || method=="ACK" || method=="MESSAGE" ))
{
log(1,"Prepaid-User");
rewritehost("prepaid.example.com");
forward(prepaid.example.com,5065);
break;
};
Best Regards
laksman
I am behind a firewall that [ports bookseller.
I use the serweb for neviar a mensage to an user and
it leaves an error:
408 Request Timeout.
and in the log it leaves this.
9(4027) fifo_get_method: method: 'MESSAGE'
9(4027) DEBUG: fifo_get_ruri:
'sip:walter@cwafrica.com.pe'
9(4027) DEBUG: fifo_get_nexthop: next hop empty
9(4027) fifo_get_headers: headers: From:
sip:rbolivar@cwafrica.com.pe
To: <sip:walter@cwafrica.com.pe>
p-version: Web_interface_Karel_Kozlik-0.9
Contact: <sip:daemon@cwafrica.com.pe>
Content-Type: text/plain; charset=UTF-8
9(4027) parse_headers: flags=-1
9(4027) end of header reached, state=9
9(4027) DEBUG: get_hdr_field: <To> [30];
uri=[sip:walter@cwafrica.com.pe]
9(4027) DEBUG: to body [<sip:walter@cwafrica.com.pe>
]
9(4027) DEBUG: fifo_uac: parse_headers succeeded
9(4027) fifo_get_body: body: sdas
9(4027) end of header reached, state=8
9(4027) DEBUG: get_hf_block: one more hf processed
9(4027) DEBUG: get_hf_block: one more hf processed
9(4027) DEBUG: get_hf_block: one more hf processed
9(4027) DEBUG: fifo_uac: EoL -- proceeding to
transaction creation
9(4027) DEBUG: mk_proxy: doing DNS lookup...
9(4027) get_record: lookup(_sip._udp.cwafrica.com.pe,
33) failed
9(4027) sip_resolvehost: no SRV record found for
cwafrica.com.pe,
trying 'normal' lookup...
9(4027) DEBUG: dlg2hash: 23003
9(4027) DEBUG: add_to_tail_of_timer[4]: 0x2ad72a80
9(4027) DEBUG: add_to_tail_of_timer[0]: 0x2ad72a94
9(4027) INFO: fifo_server: command empty
10(4052) DEBUG: timer routine:4,tl=0x2ad72a80
next=(nil)
10(4052) DEBUG: retransmission_handler : request
resending
(t=0x2ad72970, MESSAGE s ... )
10(4052) DEBUG: add_to_tail_of_timer[5]: 0x2ad72a80
10(4052) DEBUG: retransmission_handler : done
10(4052) DEBUG: timer routine:5,tl=0x2ad72a80
next=(nil)
10(4052) DEBUG: retransmission_handler : request
resending
(t=0x2ad72970, MESSAGE s ... )
10(4052) DEBUG: add_to_tail_of_timer[6]: 0x2ad72a80
10(4052) DEBUG: retransmission_handler : done
10(4052) DEBUG: timer routine:6,tl=0x2ad72a80
next=(nil)
10(4052) DEBUG: retransmission_handler : request
resending
(t=0x2ad72970, MESSAGE s ... )
10(4052) DEBUG: add_to_tail_of_timer[7]: 0x2ad72a80
10(4052) DEBUG: retransmission_handler : done
10(4052) DEBUG: timer routine:7,tl=0x2ad72a80
next=(nil)
10(4052) DEBUG: retransmission_handler : request
resending
(t=0x2ad72970, MESSAGE s ... )
10(4052) DEBUG: add_to_tail_of_timer[7]: 0x2ad72a80
10(4052) DEBUG: retransmission_handler : done
10(4052) DEBUG: timer routine:0,tl=0x2ad72a94
next=(nil)
10(4052) DEBUG: FR_handler:stop retr. and send CANCEL
(0x2ad72970)
10(4052) ->>>>>>>>> T_code=0, new_code=408
10(4052) DBG: callback type 6, id 4 entered
10(4052) DBG: acc: on_missed: no uas.request, local t;
skipping
10(4052) DEBUG: local_reply: branch=0, save=0,
winner=0
10(4052) DEBUG: local transaction completed
10(4052) !!!!! ref_counter: 0
10(4052) DEBUG: fifo UAC completed with status 408
10(4052) DEBUG: fifo_callback sucesssfuly completed
10(4052) DEBUG: add_to_tail_of_timer[2]: 0x2ad729b8
10(4052) DEBUG: final_response_handler : done
10(4052) DEBUG: timer routine:7,tl=0x2ad72a80
next=(nil)
10(4052) DEBUG: timer routine:2,tl=0x2ad729b8
next=(nil)
10(4052) DEBUG: wait_handler : removing 0x2ad72970
from table
10(4052) DEBUG: delete transaction 0x2ad72970
10(4052) DEBUG: wait_handler : done
can somebody explain to me that it happens?
do I have to put rules to the servant or to configure
in another way
the sip_exprees?
I need him to work with nat.
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