6 Oct
2003
6 Oct
'03
6:50 p.m.
Jiri,
I think the issue here is that we need to be able to set different call
timers based on the destination of the call.
If we have a user subscribed to voicemail, we would want a call_timer of
8 seconds for example.
If we have a user that isn't subscribed to voicemail or a call being
placed to the pstn, we might want a 60 second call timer.
I'm sure you can see the problem... We can't fail our outbound calls
after 8 seconds, and we can't wait 60 seconds to route incoming calls to
voicemail.
Stephen
Subject: Re: [Serusers] Implementing Voicemail-Round 2- t_newtran ERROR
Gavin,
I am not exactly sure what the problem is -- indeed, SER transactions
time
out after some period of time, that's normal behaviour. It is called
C-timer in RFC3261, we call the "tm" timer "inv_fr_invite" and can
change
its value with modparam. When the timer fires, the transaction is
cancelled
and deleted from server's memory not to waste it for ever. That's it.
If you wish for-ever ringing phones, without proxy cancelling expired
transactions
you can do so under some specific circumstances. Then proxy drops
pending transactions
silently, without explicit cancellation, and the transactions may
complete statelessly.
If my memory serves, this behaviour is actually default unless any of
frequent conditions
occurs which requires stateful completion. Such conditions are SIP
forking, use of
accounting, on_failure processing, explicitly disallowing silent
transaction drop, etc.
(You simply need stateful completion for any of these functions.)
The error as displayed in your logs is caused by your attempt to create
two
transactions for a single request. You create the first transaction
context
in route[]. When failure_route is triggered, you follow failure_route[1]
and
route[3] in which you attempt to create yet another transaction context.
Also, let me remind again: VM module was designed for use in a UAS which
stands
separately from a proxy server. We never tried to mix proxy with
voicemail in
a single server instance. I will be glad to look at it as time allows,
but I doubt
it works now.
-jiri
At 11:40 PM 9/25/2003, Gavin Bensom wrote:
if(uri=~"^sip:650[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
if(uri=~"^sip:[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
if(uri=~"^sip:1[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
> rewritehostport("10.10.10.5:5060");
> log(1,"10 digit exp match w/leading 1");
> route(2);
> break;
> };
> #forward international N digit requests to gateway
if(uri=~"^sip:011[0-9]+@(mydomain\.com|10\.10\.10\.49)"){
> rewritehostport("10.10.10.5:5060");
> log(1,"international expression match");
> route(2);
> break;
> };
>/* ********** VOICEMAIL logic *************
*/
> if
(uri=~"^sip:voicemail\+@"){
> log(1,"sip:voicemail uri match");
> route(3);
> break;
> };
>/* ****** Find Aliases and Locations of
users ********* */
># It is very important to lookup "aliases" before looking up
"locations"
>
if(!lookup("aliases")){
> log(1,"Couldn't find any matching alias");
> sl_send_reply("404", "User does not
exist");
> break;
> };
> if(!lookup("location")) {
> log(1,"unable to locate user");
> route(3);
> break;
> };
> };
> # forward to current uri now; use stateful forwarding; that
> # works reliably even if we forward from TCP to UDP
> log(1,"routing at eof: should not occur for outgoing PSTN
calls");
> t_on_failure("1");
> if (!t_relay()) {
> sl_reply_error();
> };
> log(1,"eof");
>}
>route[2]{
> log(1,"route[2]:SIP-to-PSTN call routed");
> if(!t_relay()){
> sl_reply_error();
> };
>}
>route[3]{
> log(1,"route[3]: voicemail processing");
> if(method=="ACK" || method=="INVITE" ||
method=="BYE" ||
method=="REFER"){
if(!vm("/tmp/am_fifo","conference")){
if(!vm("/tmp/am_fifo","voicemail")){
>}
>failure_route[1]{
> log(1,"failure_route[1]:jump to vm: route[3]");
> route(3);
>}
>interface: eth0 (10.10.10.0/255.255.255.0)
>match: 5060
>###
>U 10.10.10.5:53667 -> 10.10.10.49:5060
> INVITE sip:6609@10.10.10.49:5060 SIP/2.0..Via: SIP/2.0/UDP
10.10.10.5:5060..From: <sip:6502188884@10.10.10.
Steve and all,
Did you ever get resolution on the issue of the timeout and
t_on_failure commands
killing or routing PSTN calls to voicemail?
I looked through the mail archives and it isn't clear what the
resolution was.
I'm running into the same problem. If I set t_on_failure
to occur after a certain timeout, outgoing PSTN calls fail after that
timeout as well.
In fact, it seems that calls fail after the timeout even if
t_on_failure isn't
set.
I've successfully gotten outgoing PSTN calls being handled by a
different
t_relay than incoming or internal network calls.
What did you do to resolve the issue?
My config.
RedHat 9.0
kernel 2.4.20-20smp
ser 0.8.11 (i386/linux)
main.c, v 1.162.2.5
Also, when a Status: 486 (busy) is encountered on the recieving party
side, or a
timeout occurs due to fr_inv_timer, I'm getting this error in
my log
Sep 25 14:07:56 jiffypop /usr/local/sbin/ser[23164]: ERROR: t_newtran:
transaction
already in process 0x422c0b38
Anyone have any ideas on what the problem is?
ser.cft and ngrep output attached.
Thanks,
G
Steve Dolloff <sdolloff(a)noc.dls.net> wrote:
Atext%2Csec%3Amail>The New Yahoo! Shopping - with improved product
search
> I did place this portion inside the myself
check
>and it still tries to transfer to vm after the time expires.
I'm puzzled -- did not you want to transfer to
vm after the time
expires?
I will try to make this clearer. I am behind an ATA with a SIP proxy of
209.242.10.153. If I call someone else registered on my domain and they
are not available, I want to go to voice mail. If I call 1-800-555-1212
from my phone, I do not want my sip proxy to reroute the call to
voicemail after 10 seconds if no one answers(or ever for that matter).
Right now if I dial 18005551212 from my handset, I see the destination
as sip:18005551212@209.242.10.153 on the server which matches to myself
and ser tries to send it to voicemail.
Someone calling into the network is not a problem. They will never hit
our server unless the destination is local.
>This is the part that I really need help with!
When the call timer
>fails, the call goes to the route[1]. How do I get it into voice mail
>from that point?
See bellow, I think that should work.
This is what I had originally, and I get the following syslog.
Sep 10 16:36:36 voip2 ser: parse error (127,37-38): Command cannot be
used in the block
Sep 10 16:36:36 voip2 ser: ERROR: bad config file (1 errors)
Sep 10 16:36:36 voip2 ser: ser startup failed
Is says that vm is not valid in the block. According the admin guide,
only certain commands can be used within a failure block. I assume that
is the problem here. If not, please let me know as this is exactly what
I want to do.
THE SAME STUFF LIKE ABOVE, YOU DON'T WANT TO
t_relay ANYTHING
if(!vm("/tmp/am_fifo","voicemail")){
t_reply("500", "SEMS
error");
};
break;
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#
# $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=8
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
sip_warning=no
#
# ------------------ module loading ----------------------------------
#
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
#
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
loadmodule "/usr/local/lib/ser/modules/pa.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
#
#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
#loadmodule "/usr/local/lib/ser/modules/uri.so"
#loadmodule "/usr/local/lib/ser/modules/group.so"
#
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
#
# ----------------- setting module-specific parameters ---------------
#
# -- usrloc params --
#
#modparam("usrloc", "db_mode", 0)
#
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
#
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
#
#
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 2)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_fmt", "fimos")
#
#modparam("tm", "fr_inv_timer", 15) #INVITE timeout
#modparam("tm", "fr_timer", 5) #negative INVITE reply or no
#final reply for a request for ACK
#
modparam("voicemail", "db_url",
"sql://ser:heslo@localhost/ser")
#
#modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
#modparam("acc", "db_flag", 2)
#modparam("acc", "db_missed_flag", 2)
#
# ------------------------- request routing logic -------------------
#
# main routing logic
#
alias=10.10.10.49 #sip server IP address
alias=jiffypop #sip server name
alias=mydomain.com #sip domain/realm
alias=jiffypop.mydomain.com #sip server FQDN
#
route{
log(1,"entering main route");
#prevent strangers from claiming to belong to our domain;
#if sender claims to be in our domain in From header field,
#better authenticate him
# code not inserted yet :)
# 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 (len_gt( 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;
};
setflag(2); #set flag for accounting
# if the request is for other domain use UsrLoc
# (in case it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# digest authentication
log(1,"request for registration");
if (!www_authorize("mydomain.com",
"subscriber")) {
www_challenge("mydomain.com", "0");
break;
};
save("location");
break;
};
/* ********** Dial out to PSTN logic ************* */
#forward numerical 7 digit requests to gateway
if(uri=~"^sip:[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
rewritehostport("10.10.10.5:5060");
log(1,"7 digit expression match");
route(2);
break;
};
# strip 650 and forward to GW if user dials 650 before
phone no.
strip(3);
rewritehostport("10.10.10.5:5060");
log(1,"650 area code dialed, 650 stripped");
route(2);
break;
};
#forward numerical 10 digit requests to gateway, append
a 1 first
prefix("1");
rewritehostport("10.10.10.5:5060");
log(1,"10 digit expression match, prefix 1");
route(2);
break;
};
#forward numerical 11 digit requests that start with a
1 to GW
log(1,"1st if entered in route[3]
*vm*");
if(t_newtran()){
t_reply("100","Trying -- just a second");
if(method=="INVITE" || method=="REFER"){
log(1,"route[3]:method==INVITE ||
REFER");
> if(uri =~ "conference" ){
log(1,"could not
contact conference server");
t_reply("500","could
not contact conference server");
> };
> }
> else if (uri =~"echo"){
>
if(!vm("/tmp/am_fifo","echo")){
log(1,"could not
contact echo");
t_reply("500","could
not contact echo");
> };
> }
> else{
log(1,"vm module called
and failed");
t_reply("500",
"voicemail error");
};
};
break;
};
if(method=="BYE"){
log(1,"vm end/refer - begin");
if(!vm("/tmp/am_fifo","bye")){
log(1,"could not contact the
media
server");
t_reply("500" , "could not
contact the media server");
};
break;
};
}
else{
log(1,"route[3]:could not create new
transaction");
sl_send_reply("500",
"could not create new
transaction");
> };
> log(1,"route[3]:end of first method== check");
> };
5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>..Date: Mon, 01 Mar
1993 22:22:15 GMT..Call-ID:
A74AE3D-15AA11
CC-8071812E-8BB375E1@10.10.10.5..Supported:
timer,100rel..Min-SE:
1800..Cisco-Guid: 175259037-363467212-215
4725678-2343794145..User-Agent:
Cisco-SIPGateway/IOS-12.x..Allow:
INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO..CSeq: 101
INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:6502188884@1
0.10.10.5>;party=calling;screen=yes;privacy=off..Timestamp:
731024535..Contact:
<sip:6502188884@10.10.10.5:5
060>..Expires: 180..Allow-Events:
telephone-event..Content-Type:
application/sdp..Content-Length: 185....v=0
..o=CiscoSystemsSIP-GW-UserAgent 6694 6444 IN IP4
10.10.10.5..s=SIP
Call..c=IN IP4 10.10.10.5..t=0 0..m=audi
o 18640 RTP/AVP 0..c=IN IP4 10.10.10.5..a=rtpmap:0
PCMU/8000..a=ptime:20..
#
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
10.10.10.5:5060..From: <sip:6502188884
@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1(a)10.10.
10.5..CSeq: 101 INVITE..Server: Sip EXpress router
(0.8.11
(i386/linux))..Content-Length: 0....
#
U 10.10.10.49:5060 -> 10.10.10.189:5060
INVITE sip:esavelle@10.10.10.189 SIP/2.0..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..Via: SIP
/2.0/UDP
10.10.10.49;branch=z9hG4bKaa33.8c7a9b23.0..Via: SIP/2.0/UDP
10.10.10.5:5060..From:
<sip:6502188884
@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>..Date: Mon,
01 Mar 1993 22:22:15 GMT..Call-ID: A74A
E3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..Supported:
timer,100rel..Min-SE: 1800..Cisco-Guid: 175259037-363
467212-2154725678-2343794145..User-Agent:
Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, AC
K, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY,
INFO..CSeq: 101
INVITE..Max-Forwards: 5..Remote-Party-ID: <sip:65
02188884@10.10.10.5>;party=calling;screen=yes;privacy=off..Timestamp:
731024535..Contact: <sip:6502188884@10
.10.10.5:5060>..Expires: 180..Allow-Events:
telephone-event..Content-Type: application/sdp..Content-Length:
185....v=0..o=CiscoSystemsSIP-GW-UserAgent 6694 6444
IN IP4
10.10.10.5..s=SIP Call..c=IN IP4 10.10.10.5..t=0
0..m=audio 18640 RTP/AVP 0..c=IN IP4
10.10.10.5..a=rtpmap:0
PCMU/8000..a=ptime:20..
#
U 10.10.10.189:5060 -> 10.10.10.49:5060
SIP/2.0 100 trying..Via: SIP/2.0/UDP
10.10.10.49;branch=z9hG4bKaa33.8c7a9b23.0..Via: SIP/2.0/UDP 10.10.10.5
:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..From:
<sip:6502188884@10.10.10.5>;tag=4CCDE
44-B57..To: <sip:6609@10.10.10.49>..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE
..User-Agent: Grandstream SIP UA
1.0.3.81..Content-Length: 0....
#
U 10.10.10.189:5060 -> 10.10.10.49:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP
10.10.10.49;branch=z9hG4bKaa33.8c7a9b23.0..Via:
SIP/2.0/UDP
10.10.10.5:5060.
.Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..From:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57
..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-08924feebae5..Call-ID
: A74AE3D-15AA11CC-8071812E-8B
B375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent:
Grandstream SIP UA
1.0.3.81..Content-Length: 0....
##
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 500 could not create new transaction..Via: SIP/2.0/UDP
10.10.10.5:5060..From: <sip:6502188884@10.10
.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c9a7..Call-I
D: A74AE
3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..Server:
Sip EXpress router (0.8.11 (i386/linux))
..Content-Length: 0....
#
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
#
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
#
U 10.10.10.5:53667 -> 10.10.10.49:5060
ACK sip:6609@10.10.10.49:5060 SIP/2.0..Via: SIP/2.0/UDP
10.10.10.5:5060..From:
<sip:6502188884@10.10.10.5>;
tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c9a7..Date:
Mon, 01 Mar 199
3 22:22:15 GMT..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..Max-Forwards:
6..Content-Length: 0..
CSeq: 101 ACK....
#
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
###
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
#####
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
###########
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
###########
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
########
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
######
U 10.10.10.49:5060 -> 10.10.10.5:5060
SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route:
<sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F
rom:
<sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To:
<sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089
24feebae5..Call-ID:
A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101
INVITE..User-Agent: Grandstream
SIP UA 1.0.3.81..Content-Length: 0....
############################exit
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--
Jiri Kuthan http://iptel.org/~jiri/
7 Oct
7 Oct
8:42 p.m.
New subject: [Serusers] Implementing Voicemail-Round 2- t_newtran ERROR
At 05:50 PM 10/6/2003, Steve Dolloff wrote:
Jiri,
I think the issue here is that we need to be able to set different call
timers based on the destination of the call.
If we have a user subscribed to voicemail, we would want a call_timer of
8 seconds for example.
If we have a user that isn't subscribed to voicemail or a call being
placed to the pstn, we might want a 60 second call timer.
I'm sure you can see the problem... We can't fail our outbound calls
after 8 seconds, and we can't wait 60 seconds to route incoming calls to
voicemail.
I see the problem now -- sorry it took me so long. The bad news is we don't
support variable timers yet (that has something to do with a feature versus
speed trade-off). It stands actually on the upper parts of our to-do-list.
If any volunteering implementor needs it very quickly, we can provide guidance
on how we think it should be implemented.
-jiri
7723
days inactive
7724
days old
1 comments
2 participants
participants (2)
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Jiri Kuthan -
Steve Dolloff