Thanks for you help.
below are what I did in * extensions.conf
[globals]
SERADDRESS=@IP ser server:5060
[serserver]
exten => _3XXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)
exten => _3XXX,2,WaitMusicOnHold,15
exten => _3XXX,3,VoicemailMain(u${EXTEN:2})
In sip.conf, I just add
[general]
autocreatepeer=yes
Phones are ringing but I could not get voicemail.
Thanks in advance
At 13:36 18/11/2004, Yair Hakak wrote:
>hello,
> set autocreatepeer=yes in sip.conf and you should be fine.
> obviously you need to make sure that things are set up properly in
>extensions.conf to connect calls properly.
>
>
>On Thu, 18 Nov 2004 13:32:26 +0000, Ahmed Boreau <ahmed.boreau(a)esmt.sn> wrote:
> > Hi,
> >
> > I need help. I'm actually trying to set up ser+asterisk which are actually
> > working separately.
> > By now, I want to let asterisk receiving ser calls.
> >
> > I add these commands into ser.cfg
> >
> > if(method=="INVITE"){
> > if (uri=~"^sip:1[0-9]{10}@*") {
> > log(1,"Forwarding to Asterisk\n");
> > rewritehostport("10.0.0.13: 5061");
> > t_relay();
> > break;
> > }
> > }
> >
> > What could I need to do into sip.conf at * side.
> >
> > Thanks in advance
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
HI all,
Anyone who's deployed SER and Asterisk together, just wanted to
know that what are difference in functionalities between SER and
Asterisk. I mean people generally use SER as front end to do NAT and
stuff. Asterisk is generally used behind SER to maintain the dialplan
and appropriate routing. I was wondering if Asterisk can to Natting and
SER can be used as a registrar to store information, why cant we use
them interchangeably? Is there any specific killer function these two
have which are different?
Thanks,
Hitesh.
All,
This is my first post to this list so go easy on me. :-) I'm
rather new to Ser, in fact I just installed it for the first time early
in the week. I'm working on the NAThelper module to get traversal
working, I have outbound (sip phone -> NATout -> ser) working just
peachy, RTP works in both directions hooray. The question is I'm having
problems getting RTP inbound, the ring of course goes through, and RTP
from the NAT'd side of course works fine however getting back through
the NAT (from outside) for RTP in this sense fails. Let me explain the
setup:
I'm using the registrar, NAThelper, usrloc, and of course (Portaone's)
RTPproxy modules. The current SIP phone is an SNOM (yes yes, I know..).
The "endpoint" is Asterisk. When I do a sip debug on Asterisk, I see the
RTP request however it's coming from the NAT'd fake address:
v=0
o=root 780961119 780961119 IN IP4 192.168.1.101
s=call
c=IN IP4 192.168.1.101
t=0 0
m=audio 10004 RTP/AVP 0
a=rtpmap:0 pcmu/8000
a=sendrecv
I have an idea of what to fix just not sure how to fix it. Obviously we
need it to goto RTPproxy, since this is "backwards" how would I get it
to recognize the correct IP?
See my config below, most of it is ripped off of the NAThelper.cfg
example. :-) Thanks all..
NOTE: All calls are destined for ${SIPDOMAIN}, in this case, the
machines hostname. This is normal and intentional :-)
# ---- SNIPPAGE ----
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind
NAT
# 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;
};
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
log("LOG: Caught uac test 3 \n");
if (method == "REGISTER" || ! search("^Record-Route:"))
{
log("LOG: Someone trying to register from private
IP, rewriting\n");
# This will work only for user agents that support
symmetric
# communication. We tested quite many of them and
majority is
# smart enough to be symmetric. In some phones it
takes a configuration
# option. With Cisco 7960, it is called
NAT_Enable=Yes, with kphone it is
# called "symmetric media" and "symmetric
signalling".
fix_nated_contact(); # Rewrite contact with source
IP of signalling
if (method == "INVITE") {
log("LOG: fix nated sdp\n");
fix_nated_sdp("1"); # Add direction=active to
SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
};
# 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
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
if (uri==myself) {
if (method=="REGISTER") {
log("LOG: Caught register, registering user in
local db\n");
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
log("LOG: Caught uri myself\n");
# native SIP destinations are handled using our USRLOC
DB
#if (!lookup("location")) {
# sl_send_reply("404", "Do what now");
# break;
#};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&& !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
log("LOG: Caught NAT flag 6 forcing rtp proxy\n");
force_rtp_proxy();
};
if (method=="REGISTER") {
break;
log("LOG: Caught Register down in our call routing,
breaking\n");
};
#### Below is mostly my own doing ####
if (method=="INVITE") {
log("LOG: Caught INVITE \n");
if (lookup("location")) {
log ("LOG: Caught registered invite, sending
there\n");
# NOTE forcing rtp maybe bad idea for ALL users,
this is
# a quick fix (which doesn't work anyway!)
#force_rtp_proxy();
#forward(uri:host, uri:port); #nor does this
t_relay();
break;
} else if (uri=~"^sip:[0-9]*@") { # ... forward to
asterisk;
forward(xxx.xxx.xxx.xxx, 5060);
log("LOG: Tapping rowlf\n");
break;
};
};
#### ####
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
hi,
i have problem with latest version... see bellow...
version: ser 0.8.99-dev15
Nov 16 09:22:59 malignux /usr/local/sbin/ser[14609]: error:
mediaproxy/getContactURI(): error parsing Contact body
Nov 16 09:22:59 malignux /usr/local/sbin/ser[14609]: error:
mediaproxy/getContactURI(): error parsing Contact body
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: val2str()
Destination buffer too short
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: print_values():
Error while converting value to string
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: submit_query():
Column count doesn't match value count at row 1
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: mysql::insert_row
(): Error while submitting query
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: db_insert_ucontact
(): Error while inserting contact
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: insert_ucontact():
Error while inserting in database
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: insert(): Error
while inserting contact
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: contacts(): Error
while inserting record
Nov 16 09:23:00 malignux /usr/local/sbin/ser[14609]: error:
mediaproxy/getContactURI(): error parsing Contact body
Best regards, pedro
Dear ser users,
I'm configuring the jabber module these days.and find the subscribe function which located in the config file attached with jabber module. My ser version is 0.8.14.
Would anyone please tell me where I can find the function?
I searched all the modules and can't find such a function.
Thanks in advance.
Sincerely,
Kun
Hi there,
I'm trying to compile ser-0.8.14 on a Fedora Core 3 machine and I get a
bunch of error messages
[root@sipper ser-0.8.14]# make prefix=/ all
Makefile.rules:81: action.d: No such file or directory
Makefile.rules:81: crc.d: No such file or directory
Makefile.rules:81: data_lump.d: No such file or directory
Makefile.rules:81: data_lump_rpl.d: No such file or directory
Makefile.rules:81: dprint.d: No such file or directory
Makefile.rules:81: dset.d: No such file or directory
Makefile.rules:81: error.d: No such file or directory
Makefile.rules:81: fifo_server.d: No such file or directory
Makefile.rules:81: flags.d: No such file or directory
Makefile.rules:81: forward.d: No such file or directory
Makefile.rules:81: hash_func.d: No such file or directory
Makefile.rules:81: ip_addr.d: No such file or directory
Makefile.rules:81: main.d: No such file or directory
Makefile.rules:81: md5.d: No such file or directory
Makefile.rules:81: md5utils.d: No such file or directory
Anyone have any ideas on how to fix this ?
TIA
Hi Jan and List,
Could any one give me some more information about the hardware
configuration of the machine that you ran the tests on? (Speed,
memory). We are hesitating between using SER or CIRPACK
(www.cirpack.com) for our SIP services. CIRPACK was chosen mainly for
interconnection with the PSTN but it can also support SIP. Now we have
two options:
1. Use SER for all sip services and only use CIRPACK for PSTN calls
2. Use CirPack for both.
We prefer the first solution but we don't have enough information
about SER's performance. Any helps will be highly appreciated.
Nam
========================================
On Wed, 26 Mar 2003 13:31:13 +0100 Jan Janak wrote:
Hello,
On 26-03 14:00, Valery Shampal wrote:
> Hi, Jan
>
> Thank you very much indeed.
> So quick answer :-)
>
> As from the page http://www.iptel.org/ser/ under SIP Express Router (ser)
> topic
>
> ========================================================================
> Technical Information:
>
> C-Written. Ported to Linux (PC, IPAQ), BSD (PC) and Solaris (Sun).
> Throughput thousands of calls per second (CPS) on a dual-CPU PC (capacity
> needed to cover Bay Area) and hundreds of CPS on Compaq IPAQ. Support for
> both IPv4 and IPv6. Small footprint size: 300k core, all common modules
> (optional) up to 630k.
> ========================================================================
>
> A dual CPU Pc is mentioned. This was a "trigger" to ask the questions.
Yes, we use a dual Athlon CPU for performance measurements. On this HW
ser with simple configuration is able to do ~ 5000 CPS. The 5k CPS are
stateful, stateless ser could do more.
> We will use it within some Test-Demo Lab along with Hammer, Hammer ST and
> PacketSphere
> products from Empirix (http://www.empirix.com). We are their value added
> distributors here in Israel.
Unfortunately I don't know the products.
> So it might be thousands calls over IP in this Lab. As I understood from
> your answer, there are no
> firm limitations on PC hardware. In other words, one with 450MGhz CPU and
> 512MB
> physical memory might be enough. Am I right?
>
No, there is no HW manufacturer limitation. HW configuration really
depends a lot on the test scenarios. If you are going to have many
concurrent transactions, you will need at least 4 kB of memory per
transaction. So if you know duration of the transactions you can
easily calculate how fast you will run out of memory.
If you are going to use user location and will have many users
registered simultaneously, you will need some additional memory for
user location records (~2kB per record).
Our tests showed that usually memory is the bottleneck. The more
memory you have the longer your tests can run.
> So what about a dual CPU above?
That's a machine we use for testing.
Could you, please, provide us with more information regarding the
testing ? We are also interested in such testing (especially if
you can generate really high number of CPS or messages per
second). Maybe we could provide you with some ser optimizations so
both sides could benefit from it.
Jan.
Dear sir users,
I've installed a jabberd-1.4.3 server on my rh9 and a ser 0.8.14.
At present, I want to send message to MSN.
I've tried my best to follow the example xxjab.cfg attached with the jabber module.
But my gaim always fails to login my system when I choose the jabber client.
Would anyone like to give me some hint on that?
Thanks in advance.:-)
Sincerely,
Kun
Hi, Sir
I had installed the SER and working fine with the version 0.8.14.
Now I want to try to install the SERWEB package for my SER.
But I couldn't find out the location for the downloading.
Could you please provide me the info about where I could download the
SERWEB package for my ser...?
Thanks for your helping.
Thanks
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