I've been attempting to implement some call forwarding (noans, busy)
logic into my SER script without any luck
So far i've tried CPL (which doesn't seem to work at all for call
forward busy and apparently doesn't work properly with no answer)
(I must say, CPL seems like an excellent feature. Its really a shame
that many of its useful functions aren't yet supported in SER)
I've also tried setting the "fr_timer_avp" TM parameter from an AVP
database and trying to forward the call once the timer expires, so far
this has proven fruitless.
I'm curious if anyone has found a solution to this problem, or if it
possible.
Hello,
I am new member of the alias. I have installed SER with MySQL and did
some user provisioning. Where can I get SIP softphones or IM clients
that are compatible with SER.
Please adivice.
Prameet Chhabra
Hi all,
First of all two common anonymization questions:
Is it save for anonymization to alter the Contact-Header in SER if
record_route() is used to strip the username/IP? If not, how do you
accomplish this correctly?
And is there a way to modify the SDP body to change the o-Field to
change/remove the IP of the owner/creator?
Then an uac issue:
The uac module (backported to ser-0.9 with "MANUAL RESTORE" mode) works
well for me in this scenario:
The group "clir" indicates anonymization, and the config looks like this:
modparam("uac", "from_restore_mode", 2)
route {
# ...
if(is_user_in("From", "clir")) {
setflag(7);
uac_replace_from("Anonymous", "sip:anonymous@my.domain");
}
t_on_reply("1");
t_relay();
}
onreply_route[1] {
if(isflagset(7))
uac_restore_from();
}
When A calls B and A is in group "clir", the From-Header sent to B is
anonymized, and responses to A are restored correctly.
But when B hangs up, the UAC of B (Cisco-ATA186 or Mediatrix 2102)
changes From and To in the BYE message, so A receives the anonymized
From-Header as To-Header.
This is because uac_restore_from() only restores the From-Header (as the
name of the function says ;o) ), not the To-Header if there is a
vsf-Parameter available. This isn't correct behaviour, is it?
Also the Cisco-ATA strips off the vsf-Parameter in the BYE-Message so it
isn't available anyhow, but this seems to be a Cisco-Bug!?
Any Comments/Ideas?
Andy
> looks like the LCS doesn't find a matching entry in its
> "routing table".
mydomain.it is its native domain. All messages come in with
user(a)mydomain.it.
How is it possible?
DV
Hi All,
For the 0_9_0 branch, what is the 'best practice' for implementing
parallel forking?
We used to be able to point one alias at several accounts, but that
feature has regressed.
Thanks,
-Jev
Hello to all... I am nascent and I need aid because it
corresponds to me to test with the SER and Linux, and
do not have idea of as one settles and it forms, I
believe that I did was to download the SER-0.8.14, but
not like installing it, forming it and to make
tests... If somebody knows help me please!
NOTE: Excuse me, I don't speaking english...
Atentamente...
Baudilio O. Zapata G.
Teléfono: 0414-1213474
e-mails: sistemasbozg(a)yahoo.es
sistemasbozg(a)cantv.net
baudilio25(a)hotmail.com
______________________________________________
Renovamos el Correo Yahoo!: ¡250 MB GRATIS!
Nuevos servicios, más seguridad
http://correo.yahoo.es
Hello to all... I am nascent and I need aid because it
corresponds to me to test with the SER and Linux, and
do not have idea of as one settles and it forms, I
believe that I did was to download the SER-0.8.14, but
not like installing it, forming it and to make
tests... If somebody knows help me please!
NOTE:I don't speaking spanish...
Atentamente...
Baudilio O. Zapata G.
Teléfono: 0414-1213474
e-mails: sistemasbozg(a)yahoo.es
sistemasbozg(a)cantv.net
baudilio25(a)hotmail.com
______________________________________________
Renovamos el Correo Yahoo!: ¡250 MB GRATIS!
Nuevos servicios, más seguridad
http://correo.yahoo.es
Hello to all... I am nascent and I need aid because it
corresponds to me to test with the SER and Linux, and
do not have idea of as one settles and it forms, I
believe that I did was to download the SER-0.8.14, but
not like installing it, forming it and to make
tests... If somebody knows help me please!
NOTE:I don't speaking spanish...
Atentamente...
Baudilio O. Zapata G.
Teléfono: 0414-1213474
e-mails: sistemasbozg(a)yahoo.es
sistemasbozg(a)cantv.net
baudilio25(a)hotmail.com
______________________________________________
Renovamos el Correo Yahoo!: ¡250 MB GRATIS!
Nuevos servicios, más seguridad
http://correo.yahoo.es
Hi,
I'm setting up SER to act as a proxy between Asterisk and Microsoft LCS (to
enable TCP2UDP translation needed to speak with Microsoft LCS).
I'd like to keep configuration the simplest as possible.
Unfortunately, LCS refuses INVITE with a:
<504: Server Timeout>
as you can see in following output from 'ngrep' (some security sanityzing
performed):
#######
T xxx.yyy.64.118:32834 -> 10.99.1.9:5060 [AP]
INVITE sip:myuser@mydomain.it SIP/2.0.
Max-Forwards: 10.
Record-Route:
<sip:myuser@xxx.yyy.64.118;transport=tcp;r2=on;ftag=as12c76469;l
r=on>.
Record-Route: <sip:myuser@xxx.yyy.64.118;r2=on;ftag=as12c76469;lr=on>.
Via: SIP/2.0/TCP xxx.yyy.64.118;branch=z9hG4bK1988.c780cdd1.0.
Via: SIP/2.0/UDP xxx.yyy.64.118:5061;branch=z9hG4bK184e7f66.
From: "01234564" <sip:01234564@xxx.yyy.64.118:5061>;tag=as12c76469.
To: <sip:myuser@xxx.yyy.64.118>.
Contact: <sip:01234564@xxx.yyy.64.118:5061>.
Call-ID: 3b3a9d59133196ef4dfa27836e47cea9(a)xxx.yyy.64.118.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Date: Tue, 29 Mar 2005 17:34:32 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Content-Type: application/sdp.
Content-Length: 267.
.
v=0.
o=root 14974 14974 IN IP4 xxx.yyy.64.118.
s=session.
c=IN IP4 xxx.yyy.64.118.
t=0 0.
m=audio 11980 RTP/AVP 8 3 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
#
T 10.99.1.9:5060 -> xxx.yyy.64.118:32834 [AP]
SIP/2.0 100 Trying.
Via: SIP/2.0/TCP
xxx.yyy.64.118;branch=z9hG4bK1988.c780cdd1.0;ms-received-port=3
2834;ms-received-cid=1d00.
Via: SIP/2.0/UDP xxx.yyy.64.118:5061;branch=z9hG4bK184e7f66.
From: "01234564" <sip:01234564@xxx.yyy.64.118:5061>;tag=as12c76469.
To: <sip:myuser@xxx.yyy.64.118>.
Call-ID: 3b3a9d59133196ef4dfa27836e47cea9(a)xxx.yyy.64.118.
CSeq: 102 INVITE.
Content-Length: 0.
.
##
T 10.99.1.9:5060 -> xxx.yyy.64.118:32834 [AP]
SIP/2.0 504 Server time-out.
Via: SIP/2.0/TCP
xxx.yyy.64.118;branch=z9hG4bK1988.c780cdd1.0;ms-received-port=3
2834;ms-received-cid=1d00.
Via: SIP/2.0/UDP xxx.yyy.64.118:5061;branch=z9hG4bK184e7f66.
From: "01234564" <sip:01234564@xxx.yyy.64.118:5061>;tag=as12c76469.
To: <sip:myuser@xxx.yyy.64.118>;tag=9B5EAEFA3B5764273F459C938E732306.
Call-ID: 3b3a9d59133196ef4dfa27836e47cea9(a)xxx.yyy.64.118.
CSeq: 102 INVITE.
Content-Length: 0.
.
#################################################################
Logs from LCS shows this message:
Text: Unable to route the request
Result-Code: 0xc3e91002
SIP-Start-Line: INVITE sip:myuser@mydomain.it SIP/2.0
SIP-Call-ID: 572410cc05c62bfe720cbc4c6cde6b0b(a)xxx.yyy.64.118
SIP-CSeq: 102 INVITE
Data: Non-trusted source sent an FQDN/IP that doesn't match a routing table
rule
##################################################################
This is my ser.cfg:
debug=9 # debug level (cmd line: -dddddddddd)
log_stderror=yes # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=xxx.yyy.64.118
alias="mydomain.it"
alias="xxx.yyy.64.118"
fifo="/tmp/ser_fifo"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
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;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# reject REGISTER attempts for now
if (method=="REGISTER") {
sl_send_reply( "503", "Registration Unavailable" );
break;
};
# Forward to LCS if text usernames
if (uri =~ "sip:[a-zA-Z\.\_]*@*") {
log(1, "Forwarding to LCS\n");
rewritehostport("mydomain.it");
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay_to_tcp("eco.pitagora.it", "5060")) {
sl_reply_error();
};
break;
}
# Forward to Asterisk if numerical username
if (method == "INVITE") {
if (uri =~ "sip:9[0-9]{10}@*") {
log(1, "Forwarding to Asterisk\n");
rewriteport("5061");
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay)) {
sl_reply_error();
};
break;
}
}
sl_send_reply("404", "Not Found");
}
##############################################################
Any help?
Thanks in advance...
Domenico Viggiani
Hola a todos... soy principiante y necesito ayuda
porque me corresponde hacer una prueba con el SER y
Linux, y no tengo idea de como se instala y configura,
lo unico que creo que hice fue bajar el SER-0.8.14,
pero no se como instalarlo, configurarlo y realizar
pruebas... Si alguien sabe por favor ayudenme!!!
--- Simon Miles <simon(a)SystemsRM.co.uk> wrote:
> Dear All,
>
> Paul, Greger and myself would like you to know that
> the next issue of
> 'Getting Started' is now available on the
> www.ONsip.org
> <http://www.onsip.org/> web site.
>
> This new version builds upon the last issue by
> adding NAT support.
>
> We have covered both mediaproxy AND rtpproxy and
> explain how to
> implement either. Example config files are explained
> and these can be
> downloaded from the web site as well.
>
> Many thanks for all the very supportive feedback we
> have received. We
> are doing this for the community and always welcome
> feedback on the
> document to make it more readable etc. Comments are
> welcome on the
> www.ONsip.org <http://www.onsip.org/> web site,
> goto Forums and select
> the Getting Started Feedback area.
>
> To download the document, go to the download
> section.
>
> We hope this is of benefit to the Community.
>
>
> Thanks
>
>
> Paul, Simon, Greger
>
> > _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Atentamente...
Baudilio O. Zapata G.
Teléfono: 0414-1213474
e-mails: sistemasbozg(a)yahoo.es
sistemasbozg(a)cantv.net
baudilio25(a)hotmail.com
______________________________________________
Renovamos el Correo Yahoo!: ¡250 MB GRATIS!
Nuevos servicios, más seguridad
http://correo.yahoo.es
