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 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!
$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 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
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 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
[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
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What makes a great ClueCon presentation? The tech savvy crowd that attends
ClueCon loves *technical* presentations. In general, the more technical the
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- 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 $300
discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if
you have any questions about ClueCon 2012. Also, keep in mind that the
FreeSWITCH community has a conference
call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each
Wednesday at 1PM Eastern time. This is a great opportunity to talk
about open source telephony and get to know a number folks who will be at
ClueCon 2012. Stay tuned for more news about ClueCon speakers, sponsors,
and related events!
--
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12cfs2
Hi mailing,
I installed kamailio 3.2, rtpproxy 1.2.1, callcontrol 2.0.15 on a vz,
and cdrtool 8.2.5 and freeradius 2.1.10 on another vz.
The 2 vz container have public ip address, and the UAC have private ip
address.
I want to use rtpproxy, and the following are what ps and netstat
command returns about rtpproxy:
teddy@kamailio:~$ 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 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
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 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
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
My perception is that SER is something like a kernel, but Kamailio a user-oriented distribution and ecosystem wrapper around it that serves to "practicalise" it in an applied way.
--
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
Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
>Hello,
>
>if you install from sources, there is practically everything: ser +
>kamailio together, it is up to you what modules you want to use.
>
>If you install from kamailio packages (debian, centos), you get only
>kamailio specific modules (what is located in modules/ and modules_k/).
>
>Maturity of the code is the same, I am more into kamailio specific
>modules, though, since I use them. Kamailio is traditionally more
>interactive in terms of public relations, having a clear release policy,
>with packages built every time.
>
>Also, kamailio specific modules have more into SIMPLE extensions (I mean
>here presence, xcap, ...), but this is not a problem since you can use
>the modules with the ones developed in the past by SER. So there is
>nothing different between Kamailio and SER if you install from sources.
>As mentioned in the link, there are just some inter-module dependencies,
>so when using a module, you have have to use another specific one. This
>is actually valid in the same group of modules (like in modules_k), one
>module requires another one.
>
>What really make the difference in a deployment is the database
>structure used behind. If you start from scratch, it does not matter
>probably, but people upgrading from older versions, tend to stick to
>what they have, so they continue using one or the other.
>
> From my observations on this mailing list, it is more likely you get
>people answering more often for kamailio than ser. Also, there are more
>tutorials showing how to use kamailio, including integration with
>asterisk, if that is main concern, for example:
>
> * http://kb.asipto.com/asterisk:index
>
>Cheers,
>Daniel
>
>
>
>On 4/9/12 9:43 PM, Daniel Gonzalez wrote:
>> Thanks for the link.
>>
>> If I understand correctly, both projects share the same source code,
>> and implement more or less the same functionality.
>> Is there a list of specific features which are only available in SER
>> or in Kamailio?
>>
>> Which is the most widely deployed / documented option? Which is more
>> mature / stable? Which is the option which interoperates easier with
>> Asterisk?
>>
>> Sorry for the beginner questions, but I have found no place where this
>> questions are adressed.
>>
>> On Mon, Apr 9, 2012 at 9:17 PM, SamyGo <govoiper(a)gmail.com
>> <mailto:govoiper@gmail.com>> wrote:
>>
>> :-|
>>
>> http://www.kamailio.org/w/sip-router-releases/
>>
>>
>>
>> On Mon, Apr 9, 2012 at 11:06 PM, Daniel Gonzalez
>> <gonvaled(a)gonvaled.com <mailto:gonvaled@gonvaled.com>> wrote:
>>
>> 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
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>> mailing list
>> sr-users(a)lists.sip-router.org
>> <mailto:sr-users@lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list
>> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>
>> _______________________________________________
>> 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
>
>--
>Daniel-Constantin Mierla
>Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
>http://www.asipto.com/index.php/kamailio-advanced-training/
>
>
>_______________________________________________
>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