11 Sep
2003
11 Sep
'03
12:44 a.m.
> 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;
11 Sep
11 Sep
1:17 a.m.
New subject: [Serusers] Implementing Voicemail
At 11:44 PM 9/10/2003, Steve Dolloff wrote:
I guess 1-800-555-1212 is a teleophne number behind a pstn gateway, right?
then you need first to desing a good numbering plan. e.g.:
if (uri==myself) {
if (uri="9[0-9]*@") {
# calls to IP users; voicemail is used; set up failure route here
...
} else { # if the number does not begin with 9, we route to PSTN
rewritehostport("gatyeway.somewhere.com:5060");
t_relay();
# we did not setup voicemail handling here
};
> 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.
If you want to call pstn, than set up a numbering plan, distinguish
this option in the plan from ip destinations, and grant it a different
handling (no failure_route).
Someone calling into the network is not a problem.
They will never hit
our server unless the destination is local.
I see -- that's actually a bug in SER. We only tested for remote
voicemail and never gave an intergrated version a try. It is now
fixed on CVS. Feel free to update from CVS, recompile and retry.
Sorry for the inconvenience and thanks for reporting.
-Jiri
>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 
26 Sep
26 Sep
12:40 a.m.
New subject: [Serusers] Implementing Voicemail-Round 2- t_newtran ERROR
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:
> 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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---------------------------------
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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.
if(uri=~"^sip:650[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
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
if(uri=~"^sip:[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
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
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"){
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" ){
if(!vm("/tmp/am_fifo","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{
if(!vm("/tmp/am_fifo","voicemail")){
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");
};
}
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.
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-ID: 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
3 Oct
3 Oct
5:23 p.m.
New subject: [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:
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:
--
Jiri Kuthan http://iptel.org/~jiri/
> 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.
if(uri=~"^sip:650[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
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
if(uri=~"^sip:[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
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
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"){
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" ){
if(!vm("/tmp/am_fifo","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{
if(!vm("/tmp/am_fifo","voicemail")){
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");
};
}
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.
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-ID: 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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3 comments
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participants (3)
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Gavin Bensom -
Jiri Kuthan -
Steve Dolloff