Jiri and all,

ser -V **** ser version info

version: ser 0.8.11 (i386/linux)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 1.162.2.5 2003/08/27 16:04:35 andrei Exp $
main.c compiled on 21:37:39 Sep 1 2003 with gcc 3.2

OS ************

Red Hat 9
[root@jiffypop html]# uname -a
Linux jiffypop 2.4.20-20.9smp #1 SMP Mon Aug 18 11:32:15 EDT 2003 i686
i686 i386 GNU/Linux

I tried implementing voicemail by running SER twice simultaneouly on the same box. The first is running with my ser.cfg [pid 24264 below], and the second is with a slight variation of voicemail.cfg [pid 23911 below] as posted recently by Jiri.

This is all done with the 0.8.11 major release of SER. I didn't take the files from CVS as suggested. I'll try that soon. Are there any changes in CVS which are incompatible with the 0.8.11 database configuration? Are there any other changes which I would have to be careful of with respect to compatibility?

Two instances of SER running on the same box works fairly well. I occasionally get an error via syslog which looks like this:

Oct 13 12:36:07 jiffypop Sems[23889]: Error: while getting return code from Ser

Below is an example of syslog output when a user is part of the voicemail group but has no user locations in the location database.

Oct 13 12:33:50 jiffypop /usr/local/sbin/ser[24264]: no location-uri is in vm group
Oct 13 12:33:50 jiffypop /usr/local/sbin/ser[24264]: route[3]:no user location: foward to voicemail
Oct 13 12:33:50 jiffypop ser[23911]: t_newtran created
Oct 13 12:33:50 jiffypop ser[23911]: **************** vm start - begin ******************
Oct 13 12:33:50 jiffypop ser[23911]: **************** vm start - end ******************
Oct 13 12:33:50 jiffypop /usr/local/sbin/ser[24263]: ACC: transaction answered: from="V11 Main" <si
p:info@10.10.10.49>;tag=fcad7738-6281-77bd-a9bd-0c525a6382fa, i-uri=sip:6670@10.10.10.49, method=IN
VITE, o-uri=sip:sgaudio@10.10.10.49:5090, code=200
Oct 13 12:33:50 jiffypop /usr/local/sbin/ser[24266]: no location-uri is in vm group
Oct 13 12:33:50 jiffypop /usr/local/sbin/ser[24266]: route[3]:no user location: foward to voicemail
Oct 13 12:33:50 jiffypop /usr/local/sbin/ser[24266]: ACC: request acknowledged: from="V11 Main" <si
p:info@10.10.10.49>;tag=fcad7738-6281-77bd-a9bd-0c525a6382fa, i-uri=sip:sgaudio@10.10.10.49, method
=ACK, o-uri=sip:sgaudio@10.10.10.49:5090, code=200
Oct 13 12:33:52 jiffypop /usr/local/sbin/ser[24265]: no location-uri is in vm group
Oct 13 12:33:52 jiffypop /usr/local/sbin/ser[24265]: route[3]:no user location: foward to voicemail
Oct 13 12:33:52 jiffypop ser[23909]: t_newtran created
Oct 13 12:33:52 jiffypop ser[23909]: **************** vm end - begin ******************
Oct 13 12:33:52 jiffypop ser[23909]: **************** vm end - end ******************
Oct 13 12:33:52 jiffypop /usr/local/sbin/ser[24266]: ACC: transaction answered: from="V11 Main" <si
p:info@10.10.10.49>;tag=fcad7738-6281-77bd-a9bd-0c525a6382fa, i-uri=sip:sgaudio@10.10.10.49, method
=BYE, o-uri=sip:sgaudio@10.10.10.49:5090, code=200

Below is an example of a call that gets transferred to voicemail via a failure_route [timeout or "busy" reply]. Notice that the ****vm end - begin******* and *****vm end - end ****** entries are missing. Essentially, the voicemail instance of SER isn't processing the BYE messages. I'm not sure why. I guess the other instance of SER is processing them.

Oct 13 12:29:07 jiffypop /usr/local/sbin/ser[24264]: failure_route[4]
Oct 13 12:29:07 jiffypop /usr/local/sbin/ser[24264]: route[4]:Call tried and failed: forward to voi
cemail
Oct 13 12:29:07 jiffypop ser[23914]: t_newtran created
Oct 13 12:29:07 jiffypop ser[23914]: **************** vm start - begin ******************
Oct 13 12:29:07 jiffypop ser[23914]: **************** vm start - end ******************
Oct 13 12:29:14 jiffypop /usr/local/sbin/ser[24265]: ACC: transaction answered: from=<sip:650218888
4@10.10.10.5>;tag=12FB6358-82B, i-uri=sip:6633@10.10.10.49:5060;ftag=12FB6358-82B;lr=on, method=BYE
, o-uri=sip:gnoah@10.10.10.160:5060, code=200

I think these [above log messages] types of transactions are responsible for the above mentioned SEMS error, but I'm not sure.

What I really need now is a way to deal with the timer issue. Jiri, I don't think I need multiple timers. I never want SER to timeout calls placed to the PSTN. I just want one timer which I can apply to selective transactions. Right now, the timer causes all calls to fail. I understand that I can change processing on failure, and I am doing this, but it would be very nice if the timer could be used only for specific transactions. I think this would involve setting it through a function call instead of a modparam call, but I don't know since I haven't actually looked at any code.

I've also tried to just relay PSTN calls again (re-relay) through a failure route after the timer expires, but that doesn't seem to be working too well. Is this possible?

Also, I think its not necessary to implement the "single-ser" config that is capable of running voicemail if this dual configuration works well. Of cource, the "single-ser" config would make it a little simpler for users.

Thanks,
Noah

Jiri Kuthan <jiri@iptel.org> wrote:

At 04:40 AM 10/11/2003, Gavin Bensom wrote:
>Jiri,
>
>Is it possible to run two instances of ser on the same box, one acting as the proxy and the other acting as the voicemail2email server, or are to different boxes required?

I never tried -- it should, but better verify it. Just try two different
SER's at two different port numbers. Take stable version from CVS for both
SER and voicemail -- I made some recent alignments to better deal with
scenarios like these.

>Can you comment a little more on how to get SER as a proxy to transfer messages to SER as a UAS, both on a single box [if this is possible], and/or in a two box setup.

That's script fragments for the fronting proxy that should make the
job (not verified either):

# does the user wish redirection on no availability? (i.e., is he
# in the voicemail group?) -- determine it now and store it in
# flag 4, before we rewrite the flag using UsrLoc
if (is_user_in("Request-URI", "voicemail")) {
setflag(4);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
if f (isflagset(4)) route(4);
break;
};
# if user is on-line and is in voicemail group, enable redirection
if (method == "INVITE" && isflagset(4)) {
t_on_failure("1");
};

....
route[4] {

rewritehostport("foo.bar:1234");
t_relay();
}
...
failure_route[1] {
revert_uri();
lookup("aliases");
rewritehostport("bat.iptel.org:5090");
append_branch();
t_relay();
}

>I'd really like to do all this on one box as its much more economical.

I will try to get back to you with a "single-ser" config too --
I just happened to run out of time in the past week and will
not be any better off next too weeks.

-jiri