Is it possible to make the SER 0.9 (or 2.0) to reuse the listening TCP
port , usually 5060, when it relays requests to the next
SIP element in the network.
Hi All,
I am working on implementing group/directed call pickup and it is
my understanding that this requires dialog state information. Can
someone describe the best approach to implement this when using
multiple proxies? We have multiple proxies that replicate
registration information to each other.
Is there a way to share the dialog information among multiple
proxies? or should all calls be routed through one proxy?
Also, does this require presence server?
Thanks in advance.
--
Zahid
Hello,
I have 2 openser working on 2 diffrents server.
The first is doing the full job, Registration, authentifications ....
And the second is only used for SEMS so a very simple config file.
the first is working perfectly but the second create me this night a
memory error.
He was working correcly more that 1 month and I don't think that the
problem is a overload usage because during the night I have low
traffic.
./openser[5429]: ERROR: t_newtran: new_t failed
./openser[5435]: ERROR: sip_msg_cloner: cannot allocate memory
./openser[5435]: ERROR: new_t: out of mem:
mpath="/usr/local/lib/openser/modules"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "xlog.so"
loadmodule "avpops.so"
modparam("tm", "fr_inv_timer", 15000)
modparam("tm", "pass_provisional_replies", 1)
modparam("rr", "enable_full_lr", 1)
children=60
disable_tcp=yes
disable_dns_blacklist=yes
disable_dns_failover=yes
disable_dns_blacklist=true
# modparam( "tm", "tw_append",
"info_append:hdr[Content-Length];hdr[Content-Type];msg[body]")
#modparam( "tm", "tw_append",
"info_append:Content-Length=$avp(i:70);Content-Type=$avp(i:71);body=$avp(i:72)")
modparam("tm", "tw_append",
"billing:billingid=$avp(i:67);accountid=$avp(i:68);Remote-Party-ID=$avp(i:69);Pays=$avp(i:70)")
avp_aliases="bid=i:67;acid=i:68;rp=i:69;pays=i:70"
# ------------------------- request routing logic -------------------
route{
# xlog("L_ERR","BEGIN: call [$rm] ru[$ru] fu[$fu] tu[$tu]
si[$si] ct[$ct]\n");
if (!mf_process_maxfwd_header("5")) {
sl_send_reply("483","Too Many Hops");
return;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
return;
};
sl_send_reply("100","---- trining");
record_route();
# if (loose_route()) {
# xlog("L_ERR","LOOSE_ROUTE: [$rm] ru[$ru] fu[$fu]
tu[$tu] si[$si] ct[$ct]\n");
# }
if (!t_newtran()){
sl_send_reply("500","could not create transaction");
return;
};
if (method=="INVITE"){
$avp(i:67) = $hdr(x-billingId);
$avp(i:68) = $hdr(x-accountcode);
$avp(i:69) = $hdr(Remote-Party-ID);
$avp(i:70) = $hdr(Pays);
if (uri =~ "sip:.*@") {
if (!t_write_unix("/tmp/sems_sock","billing/billing")){
log("could not contact media server\n");
t_reply("500","could not contact media server");
return;
}
return;
}
# if (method=="INFO") {
# $avp(i:70) = $cT;
# $avp(i:71) = $cl;
# $avp(i:72) = $rb;
#
# if(!t_write_unix("/tmp/sems_sock","sems/info_append")){
# log("could not contact sems\n");
# t_reply("500","could not contact media server");
# }
# }
}
if (method=="BYE" || method=="CANCEL"){
if(!t_write_unix("/tmp/sems_sock","bye"))
t_reply("500","error contacting bye");
}
return;
}
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Hi,
currently I update openser 1.2.0 to 1.2.2. I found that cpl_run.c has an
error arounf line 553:
case REASON_ATTR:
<<<<<<< .mine
reason.len=n;
=======
reason.len = n;
>>>>>>> .r2629
should be:
case REASON_ATTR:
reason.len = n;
regards
Helmut
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Is anyone aware of, or does anyone have any experience with any
SIP-based multipoint (i.e. more than 2 users) video conferencing server
software?
Ideally, of course, I'd like an H263 codec module for SEMS, but barring
that, another SIP server that I can easily mesh with SER, forwarding
video requests as need be to the video conferencing server to handle
just the video mixing.
Anyone use anything like that in their environments?
N.
Hello,
I'm trying to setup some kind of iteration through an AVP list using
recursive routing, and use an AVP to create an "index" incrementing counter,
i.e.
if(is_avp_set("$avp(counter)")){
$avp(counter) = $avp(counter) + 1;
} else {
$avp(counter) = 0;
}
then I wish to apply to an avp index likewise:
if($avp(some_avp[$avp(counter)]) == null){ # this avp list index does not
exist, stop recursive routing } else { # keep doing operations }
However this is trivial, there seems not to be any easy way of going about
this, also openser.cfg errors out when I try to nest AVP's and use an AVP
value as an index value.
xl_parse_index: error parsing format [$avp(counter)])] expecting ']'
parse error (410,59-89): unknown script variable
Any thoughts? All Appreciated, thank you!
--
Brandon Armstead
Yin,
here you have a proper configuration:
http://openser.svn.sourceforge.net/viewvc/openser/branches/1.2/examples/nat…
Just make sure you install rtpproxy and start it on the same host with
your openser.
If properly installed, openser should recognize it in the logs (found
RTP proxy, support for it activated)
Cheers,
DanB
On 8/28/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote:
> Hi Dan,
>
> In fact, I have two telephones VOIP and Openser as a SIP Proxy (in the same local network,there's no NAT.), they can register and call each other with no probleme. Now I just need the stream RTP pass by Openser, but no directely between two telephone. I have installed the rtpproxy, but I don't understand how it works exactely. And I don't know if it is nessary to install mediaproxy? This is my openser.cfg, can you give me some advice?
> Thank you very much!
>
> Yin
>
>
> -----Message d'origine-----
> De : Dan-Cristian Bogos [mailto:dan.bogos@gmail.com]
> Envoyé : mardi 28 août 2007 10:29
> À : zze-WU Yin RD-MAPS-LAN
> Cc : users openser.org
> Objet : Re: [OpenSER-Users] About RTP stream
>
> Yin,
>
> It all depends on what do u want to use it for. Can u explain a bit your scenario?
>
> DanB
>
> On 8/28/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote:
> >
> > Thank you for your help. I don't need NAT, it's just a local network. What I have to use? The Rtpproxy or Mediaproxy? And what diffrence between them?
> >
> > Yin
> >
> > -----Message d'origine-----
> > De : Dan-Cristian Bogos [mailto:dan.bogos@gmail.com] Envoyé : lundi 27
> > août 2007 18:01 À : zze-WU Yin RD-MAPS-LAN Cc : users(a)openser.org
> > Objet : Re: [OpenSER-Users] About RTP stream
> >
> > Hello Yin,
> >
> > openser by itself has no RTP capability. If you want NAT bypass solutions, check out the following softwares:
> > 1. RTPProxy - http://ftp.iptel.org/pub/rtpproxy/
> > 2. Mediaproxy: - http://ag-projects.com/MediaProxy.html
> >
> > If you are looking for some RTP related applications, you can check SEMS:
> > http://www.iptel.org/sems
> >
> > Hope this helps.
> >
> > DanB
> >
> > On 8/27/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote:
> > >
> > >
> > >
> > > Hello everyone,
> > >
> > > Now my openser is just a Sip proxy which manage the sip
> > > requests and I want to manage RTP streams by the openser, that means
> > > openser recieve RTP streams and retransmit them. How can I do?
> > >
> > > Thank you very much!
> > > _______________________________________________
> > > Users mailing list
> > > Users(a)openser.org
> > > http://openser.org/cgi-bin/mailman/listinfo/users
> > >
> > >
> >
>
>
Yin,
It all depends on what do u want to use it for. Can u explain a bit
your scenario?
DanB
On 8/28/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote:
>
> Thank you for your help. I don't need NAT, it's just a local network. What I have to use? The Rtpproxy or Mediaproxy? And what diffrence between them?
>
> Yin
>
> -----Message d'origine-----
> De : Dan-Cristian Bogos [mailto:dan.bogos@gmail.com]
> Envoyé : lundi 27 août 2007 18:01
> À : zze-WU Yin RD-MAPS-LAN
> Cc : users(a)openser.org
> Objet : Re: [OpenSER-Users] About RTP stream
>
> Hello Yin,
>
> openser by itself has no RTP capability. If you want NAT bypass solutions, check out the following softwares:
> 1. RTPProxy - http://ftp.iptel.org/pub/rtpproxy/
> 2. Mediaproxy: - http://ag-projects.com/MediaProxy.html
>
> If you are looking for some RTP related applications, you can check SEMS:
> http://www.iptel.org/sems
>
> Hope this helps.
>
> DanB
>
> On 8/27/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote:
> >
> >
> >
> > Hello everyone,
> >
> > Now my openser is just a Sip proxy which manage the sip
> > requests and I want to manage RTP streams by the openser, that means
> > openser recieve RTP streams and retransmit them. How can I do?
> >
> > Thank you very much!
> > _______________________________________________
> > Users mailing list
> > Users(a)openser.org
> > http://openser.org/cgi-bin/mailman/listinfo/users
> >
> >
>
Hi,
i'm testing ser with radius using mysql. at first without inserting the
Sip-Group on radgroupcheck, my x-lite can register.
but when i add it i always get unauthorized. is there something wrong on
the values i inserted?
on my ser.cfg REGISTER part i only have this
if (method=="REGISTER") {
if (!radius_www_authorize("mydomain.com")) {
if (isflagset(8)) {xlog("L_INFO",
"Authenticating\n");};
www_challenge("mydomain.com", "0");
break;
};
if (!save("location")) {
sl_reply_error();
};
break;
};
mysql> select * from radcheck;
+----+------------------------+-----------+----+---------+
| id | UserName | Attribute | op | Value |
+----+------------------------+-----------+----+---------+
| 1 | 1001(a)mydomain.com | Password | == | TEST1001 |
+----+------------------------+-----------+----+---------+
1 row in set (0.00 sec)
mysql> select * from radgroupcheck;
+----+-----------+-----------+----+----------------------+
| id | GroupName | Attribute | op | Value |
+----+-----------+-----------+----+----------------------+
| 1 | intl-pstn | Sip-Group | == | world-pstn |
+----+-----------+-----------+----+----------------------+
2 rows in set (0.00 sec)
mysql> select * from usergroup;
+----+------------------------+-----------+
| id | UserName | GroupName |
+----+------------------------+-----------+
| 1 | 1001(a)mydomain.com | intl-pstn |
+----+------------------------+-----------+
1 row in set (0.00 sec)
