15 Nov
2003
15 Nov
'03
10:03 a.m.
It is in the cvs:
janakj@localhost:~/tmp/sip_router/test$ ls *.fifo
transaction.fifo
janakj@localhost:~/tmp/sip_router/test$
Jan.
On 14-11 12:04, Harry Behrens wrote:
Hi Jan, Nils,
Thanx.
But...:
what do I do if the source - may it also be with you - speaks in strange words:
"File test/transaction.fifo illustrates example of use
of t_uac command (part of TM module)."
and
find $SER_SRC -name ´*.fifo´ -print
yields no answer...
;-)
Your help is appreciated,
Harry
Dr. Harry Behrens
Projektleitung BIB3R
DAI Labor - Technische Universität Berlin
Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin
Fon: +49 30 314 23383
Fax: +49 30 314 21799
Email: harry.behrens(a)dai-labor.de
http://www.dai-labor.de
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-----Ursprüngliche Nachricht-----
Von: Jan Janak [mailto:jan@iptel.org]
Gesendet: Mittwoch, 12. November 2003 17:41
An: Nils Ohlmeier
Cc: Harry Behrens; Klaus Darilion; serusers(a)lists.iptel.org
Betreff: Re: AW: [Serusers] Voicemail
Yes,
and in addition to that there are some escape commands which
say: Replace this tag with the outgoing IP address.
Jan.
On 12-11 17:35, Nils Ohlmeier wrote:
IN
Hi Harry,
no there is no documentation available. You have to read the source
code to
understand the details.
Generally SER pushes details of an incoming request through
the FIFO if
requested by the config script. An application
reads from
that FIFO and
passes a reply back through FIFO again.
May the source be with you :-)
Nils
On Wednesday 12 November 2003 17:28, Harry Behrens wrote:
> Hi Klaus,
>
> is there any documenation - save of course the source -
with regards
> to this inter-FIFO communication
protocol/mechanism?
>
> All I´ve understood so far is, that it seems one pushes SIP/SDP
> "messages" into the FIFO (s.th. like a named pipe/UNIX socket?).
>
> Also, I have seen scripts which seem to indicate that there is
> something like "application escapes", where SER
functionality can
be
> addressed via s.th like
/^:name_of_function/)
>
> Can anybody point to some technical spec/description of this
> mechanism and how to configure/script it?
>
> Thanx,
>
> Harry
>
> Dr. Harry Behrens
> Projektleitung BIB3R
> DAI Labor - Technische Universität Berlin
> Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin
>
> Fon: +49 30 314 23383
> Fax: +49 30 314 21799
>
> Email: harry.behrens(a)dai-labor.de
>
> http://www.dai-labor.de
>
> > -----Ursprüngliche Nachricht-----
> > Von: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
> > Gesendet: Mittwoch, 12. November 2003 16:13
> > An: sesha(a)iic.com; serusers(a)lists.iptel.org
> > Betreff: RE: [Serusers] Voicemail
> >
> >
> > Have you installed sems? Have you started sems?
> > (http://developer.berlios.de/projects/sems)
> >
> > This is the voicemail programm for ser. You have to
install it on
> > the same machine where ser is running
and they will communicate
> > using a FIFO.
> >
> > regards,
> > Klaus
> >
> > > -----Original Message-----
> > > From: Sesha B [mailto:sesha@iic.com]
> > > Sent: Wednesday, November 12, 2003 4:11 PM
> > > To: Klaus Darilion; serusers(a)lists.iptel.org
> > > Subject: RE: [Serusers] Voicemail
> > >
> > >
> > > I'm getting this error from debugs:
> > >
> > > 500 couldn not contact announcement server.
> > >
> > > Please let me know what it means? Is it possible to put the
> > > voicemail on the PBX? I have the voicemail configured
for this
> > > number on the PBX. Thank you.
> > >
> > >
> > >
> > > -----Original Message-----
> > > From: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at]
> > > Sent: Wednesday, November 12, 2003 9:42 AM
> > > To: sesha(a)iic.com; serusers(a)lists.iptel.org
> > > Subject: RE: [Serusers] Voicemail
> > >
> > >
> > > Usually, if the phone (user) is registered, the call will
> >
> > be forwarded
> >
> > > to the SIP phone instead of forwarding it to the voicemail.
> >
> > If a want
> >
> > > to redirect the call to the voicemail after several seconds
> > > (e.g. if no one picks up the phone), you have to use the
> > > fr_inv_timer timer
> >
> > and set up
> >
> > > script as shown in:
> > >
http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESS
> G
>
> regards,
> Klaus
>
> > -----Original Message-----
> > From: Sesha B [mailto:sesha@iic.com]
> > Sent: Wednesday, November 12, 2003 2:44 PM
> > To: Jiri Kuthan; serusers(a)lists.iptel.org
> > Subject: RE: [Serusers] Voicemail
> >
> >
> > Hi,
> >
> > I'm trying to implement voicemail. When a call is made to a
> > SIP phone, and if the SIP phone is ringing, the "ngrep port
5060" on the
> > SER shows that it
> > is connected. So, it is looking it as a successful connection
> > and hence it is not going to the answering machine. Is there a
> > roundabout solution for
> > this? Please let me know. Thank you.
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
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