>
> I called m_dump() somewhere near my REGISTER
> block but IM in msilo are not being sent to my UA. !?
A copy of your ser.cfg, or at least the section that handle your REGISTER,
would help. Output from ngrep will be even better.
>
> I would like to use presence (subscribe/notify) to
> manage missed call, message store .
> For example i have two hardphones.
> If phone1 is offline or busy notify message is sent
> to phone2.
> phone1 and phone2 are always registered in location
> table and powered.
> I don't want to use "if(!lookup("location")) ... "in
> ser.cfg to manage missed call, message store.
>
I don't know how to implement such senario with subscribe/notify. It cross
my mind that this is not doable. Someone better than me may be able to help
you.
Most of us are gearing our knowledge base on location table. If you want to
do something different, be the pioneer. Not to discourage you but IMHO,
subscribe/notify better serve the IM world than call flow.
> What can set in ser.cfg in order to find the phones
> status in ser database?
>
> Harry
>
> --- Zeus Ng <zeus.ng(a)isquare.com.au> wrote:
> > See reply inline.
> >
> > > Sorry i ever sent this mail last week without
> > reply.
> >
> > Then wait for a week or two before reposting and not
> > to cross posted.
> >
> > > I do hope somebody could help me.
> > >
> > > I set up ser + serweb + asterisk with 2 ip300
> > Polycom.
> > > ip300 Polycom supporting SUBSCRIBE/NOTIFY/MESSAGE
> > > methods.
> > > When i set one phone offline (phone is powered and registered in
> > > location table) i can't store IM in msilo table.IM is
> sent to phone.
> >
> > Add the MESSAGE method in your failure route to
> > handle this situation.
> > Something similar to:
> >
> > if (method == "MESSAGE") {
> > if (m_store("1")) {
> > t_reply("202", "Accepted");
> > } else {
> > t_reply("503", "Service Unavailable");
> > };
> > break;
> >
> > >
> > > if i switch off phone IM is stored in msilo table
> > but
> > > when I switch on the phone IM is not dumped to
> > phone.
> >
> > Have you call m_dump() somewhere near your REGISTER
> > block. This is how the
> > IM in msilo being sent to your UA.
> >
> >
> > >
> > > Can we use presence (subscribe/notify) in order to
> > > store IM and missed calls in serweb because of
> > > hardphones don't have to be switched off !?
> > >
> >
> > What exactly do you want to do? No offence but can
> > you rephrase the
> > sentence. I can hardly understand what you are
> > trying to achieve.
> >
> >
>
>
>
>
>
>
> Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de
> stockage pour vos mails !
> Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/
>
I posted this message last week. It must be too obvious because no one
answered it :-).
Could anyone tell me how to get ser-0.8.99.devxx? I really need that version
for something I need to roll out.
Thank you!
Kevin
Does SER support Session Timers??????????
Samuel.
Unclassified.
>>> Jiri Kuthan <jiri(a)iptel.org> 03/07/05 01:40PM >>>
At 11:06 PM 3/6/2005, Alistair Cunningham wrote:
>Matt,
>
>Because SER handles SIP messages, not RTP streams, it cannot 100%
accurately record CDRs. For instance, if the BYE message is lost because
a client crashed, SER has no way of knowing when the call ended or how
much it cost.
It can be fairly accurate. You just need to provision the network
carefuly and use advanced
features such as session timer.
> A BTBUA gets round this problem by having the RTP packets come to it,
so if a client crashes, the RTP stream stops and the BTBUA knows the
call has finished. The disadvantage of a BTBUA is that it adds latency
and doesn't scale well, because it needs to handle all the RTP traffic.
Indeed. That's the pain. That's why B2BUA with media relay is to be
used as rarely as possible.
-jiri
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi All,
Would anyone find it valuable to be able to set append_branches via an AVP?
I think it would be good to be able to selectively set this to 1 or 0
based on who the REGISTER message is from.
Regards,
Paul
Well, I did that and this is what I got :-)
84.33.59.8:138092893
I tried "forwards" (without reverse) and got another funky IP.
-lost
-----Original Message-----
From: Iqbal [mailto:iqbal@gigo.co.uk]
Sent: Friday, March 04, 2005 12:28 PM
To: Matt Schulte; jh(a)lohi.tutpro.com
Cc: serusers(a)lists.iptel.org
Subject: RE: [Serusers] lcr module and column type in DB
yeah pretty much,
Iqbal
On 3/4/2005, "Matt Schulte" <mschulte(a)netlogic.net> wrote:
>So, just to clear this up. You still have to reverse the ip, so:
>
>example:
>209.247.17.5
>reverses to:
>5.17.247.209
>
>which long format comes to:
>85063633
>
> Thanks .. ??
>
>-----Original Message-----
>From: Iqbal [mailto:iqbal@gigo.co.uk]
>Sent: Wednesday, March 02, 2005 2:55 PM
>To: jh(a)lohi.tutpro.com
>Cc: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] lcr module and column type in DB
>
>
>
>cheers, just what I needed..one thing though I had to reverse the
>format...not sure if I was doing it wrong or something else.
>
>Anyhow for all those with the same problem manually calculating it is
>
>IP=A.B.C.D
>
>IPlong= A*16777216 + B*65536 + C*256 + D
>
>but as mentioned above I for some reason had to do
>
>D*16777216 + C*65536 + B*256 + A
>
>Iqbal
>
>
>
>On 3/2/2005, "Juha Heinanen" <jh(a)lohi.tutpro.com> wrote:
>
>>Iqbal writes:
>>
>> > Any ideas
>>
>>yes, you have to put in ip_addr field integer representation of ip
>>address. use, for example, php's ip2long function.
>>
>>-- juha
>>
>>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hello Alexey,
Monday, March 7, 2005, 1:08:14 PM, you wrote:
ANKH> Mike Tkachuk wrote:
>>I will not agree that asterisk proxy RTP streams always, you can set
>>asterisk not to do it. But will agree that asterisk is not developed
>>for such purposes and some limitations exists (but some additional
>>features exists too).
>>
>>
ANKH> How can we do it?
ANKH> Did you mean the following?
ANKH> reinvite=yes
ANKH> can_reinvite=yes
As asterisk manual says correct name of this parameter is like canreinvite=yes
not reinvite nor can_reinvite.
--
Best regards,
~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
Mike Tkachuk, ph:380-3433-47067
YES ISP, fx:380-3433-47067
Valova 17, mike|a|yes.net.ua
Kolomyia, www.yes.net.ua
Ukraine 78200 FWD: 66518
07.03.2005
ICQ# 57698805
MSN: mike_tkachuk|a|hotmail.com
~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
I found out that the proxy_challenge work fines if the UA has a real ip or if
the there is only one UA register to the server within the network. The
problem I found out is that the proxy_challenge will only send to port 5060 or
the sip port the UA defined, but if the UA is behind NAT and the NAT port is
not the 5060, the proxy authentication won't pass to the UA, so the UA can't
dial out. Is there any method to solve this problem?
Sing
Hello,
I'm kind of having the same problem as described in this message:
http://lists.iptel.org/pipermail/serusers/2004-September/011452.html
When cancelling an INVITE for which only a 100 trying response is
received, the CANCEL request does not seem to be forwarded by SER
(version 0.8.12).
client A SER client B
-> INVITE -> -> INVITE ->
<- 100 trying
<- 100 trying
-> CANCEL -> <- 180 ringing
<- 200 ok -- no more pending branches
<- 487 Request cancelled
-> ACK ->
<- 183 Session progress
<- 183 Session progress
<- 183 Session progress
<-183 Session progress
...
To client A it seems that the INVITE is cancelled, but SER does not
forward the CANCEL request to client B, so client B thinks the INVITE is
still pending and keeps sending 183 responses (which are received by
client A).
I'm not sure when SER receives the 100 trying and 180 ringing (before or
after the cancel) but the 180 response is not received by client A.
If client A sends a CANCEL request after receiving a 180 response, the
cancel handling is done correctly.
kind regards,
Sigrid Thijs
Hi all,
I'm new with the SER, I mainly have 2 problems:
1- I can't forward the calls to a gateway (suppose a Cisco router) with the
following dial-peer
dial-peer voice 10 pots
destination-pattern 9T
prefix 101
I tried the forward (ip, 5060) but it didn't work. How can I do it
(statefull)?
2-SER is supposed to send me logs to /var/log/messages, but it's not.. HELP
!!!!!!
Marc
hello ser users;
first, please don't pay attention to my english, because i'm french :-)...
I'm using the pbx asterisk and ser as a sip proxy, to pass over nat
problems. Here's my configuration :
ipphone (192.168.0.108) --->
asterisk (192.168.0.4:5061)
SER (192.168.0.4:5060) --->
nat router...
During the call, a sip message comes from asterisk to SER, but when it
arrives at the router, it seems to be corrupted.
Here's the sip/sdp message before SER :
Session Initiation Protocol
Request line: INVITE sip:0467751647@212.94.190.153;user=phone SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.0.4:5061;branch=z9hG4bK653c2038
Route: <sip:22222@212.94.190.153;user=phone>
From: "11111" <sip:11111@192.168.0.4:5061>;tag=as5ea95a2d
To: <sip:22222@192.168.0.4>;tag=1c11534
Contact: <sip:11111@192.168.0.4:5061>
Call-ID: 5de9a6a3391006b40343cfc8370a4aa0(a)192.168.0.4
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 264
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 23042 23043 IN IP4 192.168.0.108
Owner Username: root
Session ID: 23042
Session Version: 23043
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.0.108
Session Name (s): session
Connection Information (c): IN IP4 192.168.0.108 ...
the same after being routed by SER :
Session Initiation Protocol
Request line: INVITE sip:22222@212.94.190.153;user=phone SIP/2.0
Message Header
Max-Forwards: 10
Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK5fe.bf341295.0
Via: SIP/2.0/UDP 192.168.0.4:5061;rport=5061;branch=z9hG4bK653c2038
From: "11111" <sip:11111@192.168.0.4:5061>;tag=as5ea95a2d
To: <sip:22222@192.168.0.4>;tag=1c11534
Contact: <sip:11111@192.168.0.4:5061>
Call-ID: 5de9a6a3391006b40343cfc8370a4aa0(a)192.168.0.4
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 353
Route: <sip:22222@212.94.190.153;user=phone>
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 23042 23043 IN IP4 192.168.0.108
Owner Username: root
Session ID: 23042
Session Version: 23043
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.0.108
Session Name (s): session
Connection Information (c): IN IP4
192.168.0.482.231.33.XXX192.168.0.4 ...
...as tou can see, the connection information in the sdp message is a
bit strange...
My ser.cfg :
-------------------------
check_via=yes # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
fifo="/tmp/ser_fifo"
# ------------------ 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/mangler.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.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"
# 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")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock")
# ------------------------- 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 (msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
log(1, "nouvelle trame reçue\n");
if(method=="INVITE"){
log(1, "invite route\n");
add_rport();
fix_nated_sdp("3");
fix_nated_contact();
sdp_mangle_ip("192.0.0.0/255.0.0.0", "82.231.33.XXX");
force_rtp_proxy();
t_on_reply("1");
}
if (loose_route()) {
log(1, "loose_route\n");
t_relay();
break;
}
if(method=="INVITE"){
#log(1, "trame INVITE\n");
record_route();
rewritehostport("212.94.190.153:5060");
}
log(1, "relai de la trame\n\n");
if (!t_relay()) {
log(1, "erreur de relai\n\n");
sl_reply_error();
}
}
onreply_route[1]{
log(1, "on reply\n");
if(status=~"[12][0-9][0-9]")
force_rtp_proxy();
}
----------------------------
I have the same problem with ser 0.8.14 and 0.9.0. So my question is :
is this a bug in ser, or something strange in my ser.cfg ?
Did anybody ever see something like this ???
thanks.
thibault.
