I have a prefix, and i remove using the Strip in the DB
INVITE sip:8135553333@192.168.2.77 SIP/2.0
Record-Route: <sip:99998135553333@198.168.1.170;lr=on;ftag=as1074edd2>
Via: SIP/2.0/UDP 198.168.1.170;branch=z9hG4bKa5b6.ce53.0
Via: SIP/2.0/UDP 198.168.1.180:5060;branch=z9hG4bK292133e9;rport=5060
From: "Omar" <sip:5125555555@198.168.1.180>;tag=as1074edd2
To: <sip:99998135553333@198.168.1.170>
Contact: <sip:5125555555@198.168.1.180>
__________________________
Omar
Hi,
According to the RTPproxy 1.1 release notes you can fork a stream (http://www.rtpproxy.org/wiki/RTPproxy/RELENG_1_1_0_NOTES) :
"The new C (copy) command has been introduced allowing to fork off one or both RTP streams in the session to some third-party in real time."
However, after spending an afternoon searching the Internet, I can't for the life of me find any documentation on how to use this feature. So, how do you fork a stream with RTPproxy?
Any help is greatly appreciated.
Jeremy V.
Folks,
First of all I will like to apologize before hand, since I am better with numbers than with words. Therefore I hope you all can at least understand the essence of what I am trying to communicate. ;-)
I am trying to set up an application, where I use Kamailio, asterisk, and A2Billing to process pre paid calling cards, and wholesale voip. I already have setup a kamailio service using carrier route and a2billing as a preprocessor. It works beautiful, Great work guys!!!!
The questions I have are around the design for resiliency and redundancy. I will like to use another instance of kamailio for a dispatcher function to load balance between asterisk servers, since the is a limitation and performance issues with asterisk. Now I have done some research and some testing and they only way I know I can make it work is by using 302 redirection to redirect the invites from my customers to the corresponding asterisk server. If I use a direct setup using ds_select_dst asterisk gets all "confused" since the request appears to come from the dispatcher ip address. For wholesale ip, I filter a2billing on ip address and not username. My configuration is as follows:
<customer>------><dispatcher/kamailio>------><a2billing/asterisk>-----><carrierroute/kamailio>------><carrier>
Any thoughts, comments and/or suggestions are more than welcomed!
Carlos
Hello,
anyone that still has to commit something on SVN branch 1.5
@sourceforge.net, please do it before 14:00GMT, afterwards will start
releasing 1.5.2.
Thanks,
Daniel
Hello,
I would like to assign multiple DIDs (not sure how it's called in
Kamailio) for one of my sip user.
This sip user is registering with 7000 as username, and I would like
to assign 7001, 7002 numbers to this sip user.
I have tried using dbaliases function, so I have added 7001 and 7002
as aliases for 7000.
If I call 7002, then alias is retrieved fine from db, I have no
problem with that.
But 7000 is receiving this call as 7000 (so he see 7000 as a called
number, instead of 7002).
is there a way to send "called number 7002" to 7000?
Kind regards,
Dmitri
Hi
How can we set the expires value of the REGISTER OK sent back to the clients
in a dynamic way ?
We set already a value with
modparam("registrar", "max_expires", 120)
but if the UA has been detected to be behind a NAT we want to apply a
different value.
Regards,
Pascal
Hello,
thanks for testing and reporting back.
Please cc all the time to the list, so everybody interested gets updated
about conclusions.
Daniel
On 14.07.2009 0:05 Uhr, Marco Bungalski wrote:
> Hello, Daniel!
>
> Thank you for you documentation - with this my linux-knowledge increased .-)
>
>
> I've just downloaded an testet - now Kamailio don't crash with a loaded
> dialog-module after
> a receive of SIP-message containing an empty record-route-field. So good
> luck for tomorrow :)
>
> Message now:
> Jul 13 23:59:37 [23474] ERROR:core:print_rr_body: failed to parse RR
> Jul 13 23:59:37 [23474] ERROR:dialog:populate_leg_info: failed to print
> route records
>
>
> Thanks and cheers,
>
> Marco
>
>
> -----Ursprüngliche Nachricht-----
> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
> Gesendet: Montag, 13. Juli 2009 17:05
> An: Marco Bungalski; kamailio
> Betreff: Re: AW: AW: [Kamailio-Users] preparing kamailio 1.5.2
>
> Hello Marco,
>
> On 13.07.2009 14:28 Uhr, Marco Bungalski wrote:
>
>> Hello, Daniel,
>>
>> i know how to compile, but where to download the version with the bugfix
>> included?
>>
>>
> if you fetch from svn the branch 1.5 then it has the fix. Here is a
> short description of how the versioning was done so far for kamilio
> (openser) from svn:
> - when a major version is released, a branch with the first two numbers
> in the version is created (e.g., 1.5)
> - for each minor version (including the .0 - corresponding to the major
> version) a tag is created with name of the three numbers in version
> - any fix for the major version is committed in the branch. Once a new
> minor release is out, a new tag is created. Therefore, the branch has
> all the time the most updated code for a specific major release. The
> tags are snapshots at the time of release.
>
> For the case of 1.5.x series:
> - there is one branch 1.5
> - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2.
> - the tarballs are created from a svn checkout done as exampled in the
> wiki page I sent to you.
>
> probably reading a bit about svn (subversion) will help you. Anyhow,
> next major version will use GIT.
>
> Now, just to download the latest 1.5.x version, do:
> svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5
> kamailio-1.5
>
> You need subversion package installed. The continue installation as you
> do with the tarball.
>
> Cheers,
> Daniel
>
>> Hello,
>>
>> this one should help you:
>> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn
>>
>> Cheers,
>> Daniel
>>
>> On 07/13/2009 01:30 PM, Marco Bungalski wrote:
>>
>>
>>> Hello,
>>>
>>> i am sorry to say that i've not testet it because i don't know where to
>>> download the fix and how to implementate that :-(
>>>
>>> But if you give me a link i will test it today (i can simulate the bug)
>>>
>>>
>> and
>>
>>
>>> will send you a feedback.
>>>
>>> Greetings,
>>>
>>> Marco
>>>
>>>
>>>
>>>
>>> Mit freundlichen Grüßen
>>>
>>> Marco Bungalski
>>>
>>>
>>> Marco Bungalski GmbH
>>> Traversale 5
>>> 27283 Verden
>>>
>>> Telefon: +49 4231 - 776 9999
>>> Fax: +49 4231 - 776 9998
>>> Mobil: +49 172 4204774
>>>
>>> e-mail: Marco(a)Bungalski.de
>>> web: www.bungalski.de
>>>
>>> Geschäftsführender Gesellschafter: Marco Bungalski
>>> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586
>>>
>>> -----Ursprüngliche Nachricht-----
>>> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
>>> Gesendet: Montag, 13. Juli 2009 12:32
>>> An: Marco Bungalski
>>> Cc: users(a)lists.kamailio.org
>>> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2
>>>
>>> Hello
>>>
>>> On 07/13/2009 12:28 PM, Marco Bungalski wrote:
>>>
>>>
>>>
>>>> Hello, Daniel!
>>>>
>>>>
>>>>
>>>> Will the bug "Kamailio crashes when receiving an empty
>>>> Record-Route-field with using the dialog-module" be fixed in 1.5.2?
>>>>
>>>>
>>>>
>>>> If yes: Thanks for that :-)
>>>>
>>>>
>>>>
>>>>
>>> it should be already in the svn (therefore yes). Have you tested it?
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>>
>>
>>
>
>
>
Hello,
several days ago, a new module named mi_rpc was introduced to source
tree. The goal is to be able to execute MI commands using RPC interface
from sip router core.
One big advantage of this is the executing MI commands using the sercmd
tool. sercmd is a command line interface, able to do auto completion for
commands, used to change parameters, get insights of core and modules at
runtime.
If you grabbed the source code (some guidelines at:
http://sip-router.org/wiki/migration/kamailio-3.0-config), then:
cd utils/sercmd
make
./sercmd
See the readme for more options. You need to compile and load modules_s/ctl:
http://sip-router.org/docbook/sip-router/branch/master/modules_s/ctl/ctl.ht…
Back to mi_rpc, one issue that needs to be sorted out is the output
format. Right now doing a pretty-format printing which is not suitable
for xmlrpc.
Feedback is very much appreciated, thanks,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com/