I wrote the code below to rewrite an extension to a phone number (It is
called in route[LOCATION]). This code works fine with MULTIDOMAIN
enabled but when I run it as a single domain the line $fu=$var(b); does
not seem to work.. You can see that I have several xlog lines
outputting the variable values into my /var/log/messages. $var(b) has
the correct value in it, so I do not know what is happening. I have also
included the /var/log/messages output which I redacted the phone number,
…
[View More]domain, and extension for security purposes. I am sure I am going about
this backwards, but can someone provide me some guidance as to what is
going wrong?
Thank you,
Gilbert
# Rewrite - Gilbert
route[REWRITE] {
# This section rewrites the outbound calling number so that caller id
works correctly.
#!ifdef WITH_REWRITE
# lookup an outbound number to replace the extension with
$var(b)="NO REV";
sql_xquery("ca","select number from pioutalias where
username='$fU'","ra");
# determine if a outbound number exists
if ($xavp(ra=>number)) {
$var(b)="sip:" + $xavp(ra=>number) + "@" + $fd;
xlog("L_INFO","var(b): '$var(b)'");
xlog("L_INFO","fu: '$fu'");
# Assign the outbound calling number
$fu=$var(b);
xlog("L_INFO","New fu: '$fu'");
}
sql_result_free("ra");
# see if it found a number and log if it did not
if ($var(b)=="NO REV")
xlog("L_INFO","No number found for extension: '$fu'");
#!endif
}
/var/log/messages output
Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>:
var(b): 'sip:602XXXXXXX@xxxx.com'
Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: fu:
'sip:30XXXXX@xxxx.com'
Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: New
fu: 'sip:30XXXXX@xxxx.com'
[View Less]
Greetings list,
I am experiencing a strange behavior with openser 1.3.2 running on ubuntu
10. I have a basic configuration (see bellow) and i am using Linphone for
iPad as my client. I have 2 users registered and I am able to place calls
no problem. The problem is that the calls (audio or A/V) drop after 38
seconds exactly, this behavior is pretty consistent, 38 seconds is all I
can get. There is no firewall in front of the clients.
Here is my configuration, ip addresses changed to protect …
[View More]the innocent:
http://pastie.org/private/x1ck8rxjcxv6hl44hrmqg
You can see the logs of the call here (the majority):
http://pastie.org/private/4fj5efpbsrxan8plzqvfza
Am I missing something or is there anything that needs to be changed in the
routing/configuration to achieve basic functionality?
Thank you in advance!
[View Less]
$fu is not mutable.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
"Gilbert T. Gutierrez, Jr." <mailing-lists(a)phoenixinternet.net> wrote:
>I wrote the code below to rewrite an extension to a phone number (It is
>called in route[LOCATION]). This code works fine with MULTIDOMAIN
>enabled but when I run it as a …
[View More]single domain the line $fu=$var(b); does
>not seem to work.. You can see that I have several xlog lines
>outputting the variable values into my /var/log/messages. $var(b) has
>the correct value in it, so I do not know what is happening. I have also
>included the /var/log/messages output which I redacted the phone number,
>domain, and extension for security purposes. I am sure I am going about
>this backwards, but can someone provide me some guidance as to what is
>going wrong?
>
>Thank you,
>Gilbert
>
># Rewrite - Gilbert
>route[REWRITE] {
># This section rewrites the outbound calling number so that caller id
>works correctly.
>#!ifdef WITH_REWRITE
> # lookup an outbound number to replace the extension with
> $var(b)="NO REV";
> sql_xquery("ca","select number from pioutalias where
>username='$fU'","ra");
> # determine if a outbound number exists
> if ($xavp(ra=>number)) {
> $var(b)="sip:" + $xavp(ra=>number) + "@" + $fd;
> xlog("L_INFO","var(b): '$var(b)'");
> xlog("L_INFO","fu: '$fu'");
> # Assign the outbound calling number
> $fu=$var(b);
> xlog("L_INFO","New fu: '$fu'");
> }
> sql_result_free("ra");
> # see if it found a number and log if it did not
> if ($var(b)=="NO REV")
> xlog("L_INFO","No number found for extension: '$fu'");
>#!endif
>}
>
>
>/var/log/messages output
>
>Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>:
>var(b): 'sip:602XXXXXXX@xxxx.com'
>Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: fu:
>'sip:30XXXXX@xxxx.com'
>Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: New
>fu: 'sip:30XXXXX@xxxx.com'
>
>
>
>_______________________________________________
>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>sr-users(a)lists.sip-router.org
>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[View Less]
Hi all,
I'm excited to announce the release of the Sipwise sip:provider
Community Edition v2.5, a free and open source turn-key platform, which
uses Kamailio, Sems and Asterisk in its core to provide a full-blown and
feature-rich VoIP soft-switch.
http://www.sipwise.com/news/announcements/spce-v2_5-release/
New core features are IPv6 and video support, serial call hunting and
time based routing, beside lots of other fixes and enhancements.
Check our VM images to get started within a couple …
[View More]of minutes and learn
what can be built with the power of Kamailio as a stateless
load-balancer and a stateful proxy/registrar, and Sems as an SBC and
application server.
Thanks a lot to all developers, contributors and members of this mailing
list to make this possible and for helping us out!
Andreas
[View Less]
[image: Inline image 2]
ClueCon 2012 - Call For Speakers
ClueCon - the open source telephony conference by developers, for
developers - would like to announce that we are having an open call for
speaking proposals for this year's event. If you have an idea for a
technical presentation for ClueCon 2012 then we would like to hear about.
What makes a great ClueCon presentation? The tech savvy crowd that attends
ClueCon loves *technical* presentations. In general, the more technical the
…
[View More]presentation, the better. If you are thinking about a presentation then
consider these points:
- ClueCon talks are 30 minutes in length, including Q&A time with the
audience
- ClueCon has a special focus on open source VoIP and telephony projects
like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio
- Attendees enjoy hearing about projects built with open source tools
- Highly technical discussions that show the nuts and bolts are
especially well-liked
- The audience appreciates seeing and participating in live
demonstrations
Please send your proposals to marketing(a)cluecon.com. Be sure to include a
working title, description of the talk, and name of the presenter. Don't
delay! There are a limited number of openings.
ClueCon 2012 Registration Information
ClueCon 2012 registration is now open. Visit the registration
page<http://www.cluecon.com/register?cc12cfs>for details. Be sure to
book your room at the
Wyndham <http://www.cluecon.com/hotels?cc12cfs> and qualify for the $300
discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if
you have any questions about ClueCon 2012. Also, keep in mind that the
FreeSWITCH community has a conference
call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each
Wednesday at 1PM Eastern time. This is a great opportunity to talk
about open source telephony and get to know a number folks who will be at
ClueCon 2012. Stay tuned for more news about ClueCon speakers, sponsors,
and related events!
--
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12cfs2
[View Less]
Hi mailing,
I installed kamailio 3.2, rtpproxy 1.2.1, callcontrol 2.0.15 on a vz,
and cdrtool 8.2.5 and freeradius 2.1.10 on another vz.
The 2 vz container have public ip address, and the UAC have private ip
address.
I want to use rtpproxy, and the following are what ps and netstat
command returns about rtpproxy:
teddy@kamailio:~$ ps aux | grep rtpproxy
teddy 22866 0.0 0.0 3312 800 pts/0 S+ 11:08 0:00 grep
rtpproxy
teddy 31326 0.0 0.0 11360 804 ? Ssl Apr06 …
[View More] 0:02
/usr/sbin/rtpproxy -F -l our_public_ip -s udp:localhost 22222
teddy@kamailio:~$ sudo netstat -pln | grep rtp
udp 0 0 127.0.0.1:22222
0.0.0.0:* 31326/rtpproxy
And my problem is: when I make calls, for example the UAC1 calls UAC2,
with ngrep I see the UAC1 calls UAC1 and not UAC2, and I don't know why.
In kamailio config file, there are directive WITH_NAT for everything
related to rtpproxy (loadmodule, routing logic, etc) as you can see in
the attached conf file.
I try disable the use of this directive WITH_NAT so I disable the use of
rtpproxy in kamailio and it works: when UAC1 calls UAC2, the UAC2 is
ringing.
Please tell me what am I doing wrong, or to use rtpproxy without NAT is
not possible ?
In attached file the kamailio config file, the diff between rtpproxy
enable and rtpproxy disable, and the result of ngrep and what is in
syslog when I made calls.
Thanks in advance.
--
Rabary Teddy
Inutile d'imprimer ce mail
[View Less]
Hello,
I am starting to deploy a SIP router, and after reading the
documentation in http://sip-router.org I am a bit confused. I am
planning to integrate the SIP router with an asterisk PBX. Which of
the available projects is recommended to get in touch with the
technology, SER or Kamailio? Which are most of you in the list using?
Is there a particular use case where one or the other is more
appropriate?
Thanks,
Daniel Gonzalez
Hello
I need help concerning this issue , i am creating a testing scenario
to make the RTPPROXY work , where i have a server having installed on
it Asterisk , running on (192.168.10.15), and another pc connected to
the Asterisk server having the ip (192.168.10.17) , on this same pc
i've installed RTPPROXY and managed to insert another Ethernet card
to connect to a client on ip (192.168.20.3) . After connecting any
client on this ip 192.168.20.* , i manage to make a call i hear the
ring but …
[View More]no audio conversation .
My RTPPROXY command is :
/usr/sbin/rtpproxy -F -s udp:192.168.10.17:22222 -l
192.168.20.3/192.168.10.17 -d DBUG:LOG_LOCAL0
Thank you
[View Less]
Greeting,
while Registering with correct authentication I am getting : status
407 proxy authentication required
even though the registration is successfull, do you know how I can get
rid of this error
Regards
Marwan