I would like to stop 183 Session Progress messages. I can see then
arriving and being dealt with by the onreply_route code, but how can I
stop then being forwarded by the sip server ?
Thanks
Simon
I must thank all the people who have helped me over the last few days
with my queries...Im nearly there I think!
However I am now trying to set up SerWeb. I am new to Linux (any
flavour of it)so this is proving somewhat difficult.
I am following the instruction detail on
http://www.wifi.com.ar/doc/voip/ser/ser-howto.html.html
I modified /etc/php.ini for the regsiter globals=on change. Then
according to instructions I should do the following:
Move the html directory to the root of your web server:
mv html /var/www/html/htdocs/serweb
Move the phplib directory to your web server application directory:
mv phplib /var/www/html/phplib
However these instructions are for red hat and I am running Suse
Linux 9.0. I did the following because I believe this is the root of
my webserver:
mv html /srv/www/htdocs/htdocs/serweb
mv phplib /srv/www/htdocs
Could someone who has set up serweb on suse verify that this is
correct??...I then made the remaining changes such as adding "../" to
the php files as documented.
When I run http://172.16.3.15/serweb in a browser window, I get an
error saying the requested URL /serweb was not found....
Id really appreciate if someone would tell where Im going wrong
because I am finding this difficult to troubleshoot when I am not
familiar with Linux.
Thanks as always,
Aisling.
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Hi all,
I am brand new to sip and ser. I am trying to wrap my head around all
this. I am trying to route SIP (duh) through a box with multiple
interfaces, and set the return-route to the IP of the outbound interface.
I'm trying to use set_advertised_address before any record_route(s), but
it doesn't seem to do anything. How to properly use set_advertised_address?
Thanks,
Tim
Hi all
how to use proxy_challenge to work with 2 domains instead of 1 with example
above:
if (!www_authorize("xxx.org", "subscriber")) {
proxy_challenge("xxx.org", "0");
break;
};
thks
Ok I had some issues and issued KILLALL ser
Now when I serctl start:
debian:/var/run/ser# serctl start
Starting SER : started pid(417)
When I check ps -aux I see it is not running? Did I break it?? I installed latest stable .deb files. When I first installed I saw many child process running. I ran into this same situation on Fedora Core 2. So I built a new server and loaded wih Debian.
Rebooting both servers yields same result. I cannot restart ser.
Any insight would be awesome
Eric
Hello List.
I don't know if this is the right place to ask this question, but i
want to know if someone can give me a few guidelines on how to set up a STUN
server, in particular the iptel STUN server.
As far as i know i need to Ethernet Interfaces, with different IP's each one
(in the public internet). In my case, the stun server seems to start ok,
but i have a doubt. I configured in my DNS server a entry with
stund.mydomain.com pointing to the IP of my STUN server, Which IP address
i have to configure for the domain to point?. Do i need a domain name for
my stun server or i just need an IP address?.
In my actual scenario the configuration is like this:
IP1: xx.xx.xx.36
IP2: xx.xx.xx.38
In my DNS :
stund.mydomain.com ---- xx.xx.xx.36
Is this ok?
I really hope that someone can help me here!
Thanks in advance.
Regards,
Ricardo .-
Hi All,
I have currently connected an analog phone to an AddPac ATA.
AddPac registers itself to the SER and there is noproblems with that.
The AddPac can receive calls. But when I try to make a call from the
analog phone it doesn't go through. The proxy asks for proxy
authentication. One thing I noticed is that for registration the "realm"
field contains the IP address of SER whereas the realm field in the
re-Invite contains the private IP address of the AddPAc. Is this the
problem? If you any clue how to solve it.
Thanks,
Hitesh.
Hi All.
Is this allowed in ser.cfg and if so, what causes route(1) to evaluate true or false?
route {
if (route(1)) {
.. do something
} else {
.. do something else
};
}
route[1] {
if (some_condition) {
setflag(xx);
} else {
break;
};
}
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Hi everybody,
I've read at Asterisk configuring manual to integrate SER and *, and I saw that if I have my Users registered at SER, if I want them to have voicemail at Asterisk (which is behind ser) I must the peer block configuration above at sip.conf:
[2114]
type=peer
username=2114
insecure=yes
canreinvite=no
context=default
mailbox=2114
host=SIP003094C274B3.bna01.isdn.net
This means that if I ad 500 users, I must configure 500 blocks like this on (changing of course the configuration for each user) to every user ? Is that right or did I undertand wrong ??? Doesn't it slow down the running process of asterisk ?
--
Felipe Martins
Linux System Administrator
Tep Solution Provider
Mundivox Communications
Rua Lauro Muller, 116/Sala 505
RJ - Brasil - 22290-906
Tel.: 55 21 3820-8839
Fax.: 55 21 3820-8844
