Hi
when i configured ser.cfg it show the errors please
any one send me the ser.cfg configuration file
radiusclient - 0.4.1
ser - 0.8.14 these are the versions
Please send me thank u
Sailatha
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Hello list.
Is there a default or recommendable pair-of-timers-REGISTER
settings, for SER and endpoints. I'm using Linksys, and SER, the Linksys
are PAP2 and the majority of theses endpoints are NAT'd.
I would really appreciate and advice on this.
Thanks
Ricardo Martinez.-
When I receive a SIP INVITE from my service provider, the normal call
flow is for SER to forward it to my Asterisk for authentication.
However, if the Asterisk is down, SER keeps attempting by sending
multiple INVITEs to Asterisk, until it receives a CANCEL from the
service provider.
The preferred behavior is for SER to give up after 2-3 unsuccessful
attempts and send a "503 Service Unavailable" message back to the
service provider. Is there a way to do this in ser.cfg or anywhere else
in SER? Any examples would be much appreciated.
Regards,
SCM
Sorry for a touch of a cross-post (I posted a similar quetion to
serdev), but I don't know if I should have posted to ser-users instead
in the first place. Again, my apologies.
I have previously ready posts about the CPL proxy timeout value being
ignored because of a lack of ability in the tm module. After reading
some posts about being able to set dynamic invite timers via avp, it
looks like tm should handle this type of thing fine. (Again forgive
me, I have only started sifting through the SER code this week--trying
to wrap my head around it all). Since tm can support dynamic timers,
is there a way to modify the cpl handling code to set the callee
invite timer to the value in the CPL proxy timeout?
I ask because we are currently routing all calls from endpoints
through ser, to one of several asterisk boxes to handle the dialplan
(ring phone for n seconds then do x type stuff), and then back out to
ser. Since asterisk isn't a sip proxy, you get all kinds of fun
problems when a call is routed in such a way that one leg is on one
asterisk box, and the other on another and then a transfer takes
place. We would love to be able to handle all of our dialplan stuff
in CPL, but have to be able to do things like "Call my phone for 10
seconds, then ring these 3 phones for 10 seconds if I don't answer",
etc. It looks like CPL should handle this kind of thing
perfectly--except setting the timeout for a call to roll to noanswer
based on destination. I can't be setting these rules directly in the
ser.cfg because I have lots of endpoints, and they can change how long
their phone rings etc. through a web interface.
Is this type of behaviour already supported in some other way? Is it
possible to relatively easily add the proxy timeout support to CPL by
having it set the timeout using AVP? (btw, what does that acronym
stand for?) Are there not many people that would benefit from this
type of modification (i.e. do I need to brush up on my C skills :-) )?
Any info would be greatly appreciated. Thanks!
Hi all,
I'm planning to use SER with a 3rd party billing platform that
understands Cisco VSA as described here
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/vs….
I've been looking into RADIUS implementation in SER and looks like it
cannot be used out of the box, there's no support for authorization
and native RADIUS authentication, only digest. Before I go on and add
necessary code to auth_radius module, I'd like to know if anyone has
implemented it and would like to share/sell his code.
Thanks.
Hey,
I've got all my call forwarding stuff setup in ser using the avpops
module and everything works as expected. I can dial a sequence and store
the call forwarding numbers in the database and when calls come I can
retrieve the info and change the RURI using avp_pushto. The problem is
this, and I'm wondering if I'm missing something or I need to goto our
transit gw vendor.
- 12223334444 & 12223334445 are PSTN numbers
- 13334445555 is a SIP number registered with ser
1. inbound call : 12223334444 -> 13334445555
2. ser receives the invite and checks the avpops table
3. ruri is re-written to contain 12223334445 and a route() statement
is called to forward the call back out the PSTN gateway
4. the transit gw never responds to the invite because the Cseq field
follows that of the previous invite the transit gw sent to us and
it's not expecting an invite.
5. the inbound call is finally cancelled and the transaction ends as
expected.
Should I be doing something else before I rewritehostport and forward
back to PSTN or is it something the vendor should be doing but that's
not happening?
Any insight is appreciated.
-Evan
Hi all,
my SER Proxy is crashed.
My SER version is:
ser -V
version: ser 0.9.0 (i386/linux)
flags: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_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 1.197 2004/12/03 19:09:31 andrei Exp $
main.c compiled on 10:35:08 Feb 7 2005 with gcc 3.3
and here the bt of the coredump file:
#0 0x4005c1b1 in kill () from /lib/libc.so.6
(gdb) bt
#0 0x4005c1b1 in kill () from /lib/libc.so.6
#1 0x4005bde5 in raise () from /lib/libc.so.6
#2 0x4005d5a8 in abort () from /lib/libc.so.6
#3 0x0805a3e6 in sig_alarm_abort (signo=14) at main.c:428
#4 <signal handler called>
#5 0x400f8927 in sched_yield () from /lib/libc.so.6
#6 0x402a2afa in get_lock (lock=0x404f3b58) at fastlock.h:148
#7 0x402a2874 in lock_udomain (_d=0x404f3b38) at udomain.c:438
#8 0x402a277e in timer_udomain (_d=0x404f3b38) at udomain.c:403
#9 0x4029e0c4 in synchronize_all_udomains () at dlist.c:321
#10 0x402a5d18 in destroy () at ul_mod.c:280
#11 0x08074f1b in destroy_modules () at sr_module.c:356
#12 0x08059fef in cleanup (show_status=1) at main.c:355
#13 0x0805aae3 in handle_sigs () at main.c:514
#14 0x0805c424 in main_loop () at main.c:1167
#15 0x0805d058 in main (argc=3, argv=0xbffff814) at main.c:1568
Any suggestions?
Many thanx
Verbal
Hi all
I have some problems with ser.conf, i would like to know how i can check if user exist in subscriber table.
I have some radius authentication -> This only athenticate the users.
But i would like also to check if the user exist in subscriber table before i can give him permissions to call out.
Thanks for the help.
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Hello:
I am trying determine if the called number equals the called number
in my SER 0.8.14 stable proxy. I've tried several combinations of
check_to and check_from without success. Mailing lists articles on
this topic seem to refer to features in development or 0.9 branches.
Is it possible to check for this condition in the above release? If so
a hint on how to do it would help.
Thanks,Steve
