Hi guys,
thanks to you all I have my voicemail up and running now. But as people
are greedy and always want more, i guess i have another question. Is it
possible to have multiple sip phones ring at the same time and who ever
answers first gets the call? It's for our technical support group. I
have a call coming from PSTN onto a VEGA gateway that does all the sip
conversion and talks to our ser server.
thanks.
Srbo Cvetkovic | CityNet, Inc.
srbo(a)city-net.com | Pittsburgh, PA
voice: 412.481.5406 | fax: 412.431.1315
Hi
I guess you have to wait a few more months. Next version of Cisco Call Manager is
supposed to support SIP on the "outside" I understand. Or maybe use Asterisk as
a H.323/SIP gateway? Anyone tried?
rgds,
/Staffan Kerker
Sweden
-----Ursprungligt meddelande-----
Från: Jiri Kuthan [mailto:jiri@iptel.org]
Skickat: den 26 januari 2004 10:55
Till: Raymond May; serusers(a)lists.iptel.org
Ämne: Re: [Serusers] Re: Peering SER and Cisco Call Manager
Cisco Call Manager does not speak SIP. SIP interconnection to a call
manager does not work. -jiri
At 07:25 PM 1/23/2004, Raymond May wrote:
>Hi SER Gurus,
> My SER server has been running just
>perfect for the last 9 months and I am now trying to
>connect my private VoIP network to my Employer's. They
>have a Cisco Call Manager and I am trying to figure
>out how to configure SER so I can make calls from my
>private network to users that are registered to the
>Corporate ( Cisco Call Manager ). Ideally I would like
>to be able to make calls from my private network to
>the users on the Corporate and have users on the
>Corporate be able to reach users on my VoIp Network (
>SER Server ). Any suggestions would be appreciated.
>Below is current connectivity between my private
>network and our corporate network.
>
>SER Users --- SER Server --- Internet --- Cisco Call
>Manager --- Corporate Users
>
>
>Brgds,
>Ray May
>
>__________________________________
>Do you Yahoo!?
>Yahoo! SiteBuilder - Free web site building tool. Try it!
>http://webhosting.yahoo.com/ps/sb/
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Then make sure you are using the latest CVS stable branch.
(see www.iptel.org/ser/cvs)
I'm not sure that's going to solve your problem but as a matter
of fact, older versions don't support NAT traversal extensively.
-jiri
At 05:20 PM 1/23/2004, Federico R. SANCHEZ Y NAPAL wrote:
>Hi, I'm using the rpm packge:
>
>ser-0.8.12-0.i386.rpm on Linux RedHat 9.0
>
>I tried to use a source distribution (ser-0.8.12_src.tar.gz), and got
>the same result.
>
>Regards,
>-fs
>
>-----Mensaje original-----
>De: Jiri Kuthan [mailto:jiri@iptel.org]
>Enviado el: viernes, 23 de enero de 2004 11:24
>Para: Federico R. SANCHEZ Y NAPAL; serusers(a)lists.iptel.org
>Asunto: Re: [Serusers] nathelper
>
>What SER version are you using? (Unless you use stable CVS branch,
>that's
>a very likely explanation for the problem you experience.)
>
>-jiri
>
>At 08:19 PM 1/22/2004, Federico R. SANCHEZ Y NAPAL wrote:
>>Hello: I'm using a NATed UA to terminate a call in the PSTN using a
>Cisco 5300. I want to use the RTP relay capability. In the script, I
>wrote:
>>
>> ...
>> force_rtp_proxy();
>> rewritehost("200.60.XXX.XXX");
>> forward("200.60.XXX.XXX", 5060);
>> ...
>>
>>Where 200.60.XXX.XXX is the GTW IP.
>>
>>I get an error in the log:
>> "/usr/sbin/ser[7039]: ERROR: extract_mediaip: no 'c=' in SDP"
>>
>>The SIP Message sent to de GTW has a wrong content-length header, so
>the GTW rejects it. Here I copy the message:
>>
>>INVITE sip:2345674284618@200.60.30.24 SIP/2.0
>>Record-Route: <sip:2345674284618@200.32.43.55;ftag=673501644;lr=on>
>>Via: SIP/2.0/UDP 200.32.43.55;branch=0
>>Via: SIP/2.0/UDP
>200.68.54.46:5060;rport=25127;branch=z9hG4bKFE5D22D6B6C4407884
>>2396149F64C3DE
>>From: 1001 <sip:1001@200.32.43.55>;tag=673501644
>>To: <sip:2345674284618@200.32.43.55>
>>Contact: <sip:1001@200.68.54.46:25127>
>>Call-ID: 107D459F-7402-480D-988A-FC87455E1E5D(a)192.168.0.15
>>CSeq: 31862 INVITE
>>Max-Forwards: 69
>>Content-Type: application/sdp
>>voip-gtw04#
>>User-Agent: X-Lite build 1101
>>Content-Length: 315296
>>v=0
>>o=1001 23178458 23178458 IN IP4 200.68.54.46
>>s=X-Lite
>>c=IN IP4 200.32.43.55
>>t=0 0
>>m=audio 35052 RTP/AVP 0 8 3 98 97 101
>>a=rtpmap:0 pcmu/8000
>>a=rtpmap:8 pcma/8000
>>a=rtpmap:3 gsm/8000
>>a=rtpmap:98 iLBC/8000
>>a=rtpmap:97 speex/8000
>>a=rtpmap:101 telephone-event/8000
>>a=fmtp:101 0-15
>>a=direction:active
>>I would appreciate any help about this.
>>Best Regards,
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
>--
>Jiri Kuthan http://iptel.org/~jiri/
--
Jiri Kuthan http://iptel.org/~jiri/
Hello Sebastian,
As Klaus says, ask your questions. One month ago, I started to
configure SER, and this list have helped me a lot. So, I'll be
delighted to help you with SER as much as I can.
Bye
Curro
----- Mensaje Original -----
De: "Klaus Darilion" <darilion(a)ict.tuwien.ac.at>
Fecha: Lunes, Enero 26, 2004 1:10 am
Asunto: RE: [Serusers] NEW USER...
> hi sebastion!
>
> ask your questions to the mailing list, than you get the help of
> several users und other may benefit too.
>
> regards,
> klaus
>
>
> -----Original Message-----
> From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar]
> Sent: Fri 23.01.2004 18:58
> To: serusers(a)lists.iptel.org
> Cc:
> Subject: [Serusers] NEW USER...
> Hi everybody, I am new on this of SER and I am really confused...
> I want
> to know if someone can contact me by MSN (gnocetti(a)hotmail.com)
> and give
> me some clues.... to configure and make it works. Really I tried a lot
> but never succesfully, please if someone can introduce me to this
> excelent tool, becaus I want to implement SER like a PROXY SIP and
> ROUTING... for Carrier....
>
>
> TX
>
> Sebastian.
>
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
hi sebastion!
ask your questions to the mailing list, than you get the help of several users und other may benefit too.
regards,
klaus
-----Original Message-----
From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar]
Sent: Fri 23.01.2004 18:58
To: serusers(a)lists.iptel.org
Cc:
Subject: [Serusers] NEW USER...
Hi everybody, I am new on this of SER and I am really confused... I want
to know if someone can contact me by MSN (gnocetti(a)hotmail.com) and give
me some clues.... to configure and make it works. Really I tried a lot
but never succesfully, please if someone can introduce me to this
excelent tool, becaus I want to implement SER like a PROXY SIP and
ROUTING... for Carrier....
TX
Sebastian.
yes, there is a limit. its called "max branches" or similar and can be configured before compilation in one of the config files (search the archive). the maximum is around 30.
klaus
-----Original Message-----
From: Arnd Vehling [mailto:av@nethead.de]
Sent: Fri 23.01.2004 17:47
To: Jiri Kuthan
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] multiple phone rings
Jiri Kuthan wrote:
>
> register multiple phones with the same address, that's it. -jiri
Yupp, very nice feature. Is there a limit to the number of how many
sip-clients
can subscribe with the same sip-address?
-- Arnd
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
hey. I downloaded the CVS version of sems.. and Im running FreeBSD
(uname -a )
FreeBSD voip4.7-RELEASE FreeBSD 4.7-RELEASE #0: Wed Oct
9 15:08:34 GMT 2002
root@builder.freebsdmall.com:/usr/obj/usr/src/sys/GENERIC I compile sems
with gmake I get this :
AmRequest.cpp: In method `void AmRequestUAC::execute()':
AmRequest.cpp:568: `req' undeclared (first use this function)
AmRequest.cpp:568: (Each undeclared identifier is reported only once
AmRequest.cpp:568: for each function it appears in.)
gmake[1]: *** [AmRequest.o] Error 1
gmake[1]: Leaving directory `/root/sems/answer_machine'
gmake: [all] Error 2 (ignored)
anybody know why this is happening? I need to use the CVS version of
sems , because Im running SER CVS version.
- Atle
Hey guys.
Im trying to configure the voicemail module. tho im haveing some weird
problems..
Im getting these errors:
Jan 25 11:27:17 voip ser[97088]: ERROR: vm_mod_init: unable to bind db
Jan 25 11:27:17 voip ser[97088]: init_mod(): Error while initializing
module voicemail
Tanks for the help
- Atle
Here's my config file..
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 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=7
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"
alias="voip.domain.com"
# ------------------ module loading ----------------------------------
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/vm.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"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
# modparam("usrloc", "db_mode", 0)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ----------------- setting module-specific parameters ---------------
modparam("voicemail", "db_url","mysql://ser:heslo@localhost:3306/ser")
# ------------------------- request routing logic -------------------
# main routing logic
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 (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;
};
# 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") {
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
#inserted by klaus
if (method=="INVITE") {
record_route();
force_rtp_proxy();
/* set up reply processing */
t_on_reply("1");
};
# 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();
};
# Voicemail specific configuration - begin
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if(t_newtran()){
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
log("**************** vm start - begin ******************\n");
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
log("**************** vm start - end ******************\n");
break;
};
if(method=="BYE"){
log("**************** vm end - begin ******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
log("**************** vm end - end ******************\n");
break;
};
}
else {
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
};
};
# Voicemail specific configuration - end
}
#inserted by klaus
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
if (status=~"[12][0-9][0-9]")
force_rtp_proxy();
}
