[SR-Users] Kamailio 4.0 & Asterisk realtime (Failed to import bind_ob, auth: qop set, but nonce-count ...)

Daniel-Constantin Mierla miconda at gmail.com
Mon Jun 10 09:07:48 CEST 2013


Hello,

On 6/7/13 9:23 PM, Thomas Martin wrote:
> Hello,
>
> thanks to your responses.
>
> In the meantime, I have read Olle's slides a few times trying to understand the ramifications of the different approaches outlined (I am new to kamailio and pretty new to asterisk too). Routing calls to asterisk only when needed for media services and keeping all user data in kamailio, seems to be the fitting approach for my desires.
> It seems as if the http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x explains how to setup a system like that. - However, the article refers to older versions of both, asterisk & kamailio.
> Would you still recommend to follow those same instructions but use the current releases (11.4 & 4.0.1)? - Or can you push me into a better direction ?
that wiki page, even old, still gives good overview of how to do it, it 
may require to do some adjustments to match database structures and 
config files from the latest versions of asterisk and kamailio.

Cheers,
Daniel

>
> Again, thank you very much for your help!
>
> Best regards,
>
> -Thomas
>
> ps: 	Being new to this mailing list, I am unfortunately unaware of topics that might already have been exhaustedly covered - I apologise. - Also, I decided to reinstall everything and start from scratch - at this time don't want to bother anybody with the logs that just document previously failing attempts.
>
>
> On Jun 7, 2013, at 11:33 , "Olle E. Johansson" <oej at edvina.net> wrote:
>
>> 7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla <miconda at gmail.com>:
>>
>>> Hello,
>>>
>>> first, as pointed in other related discussions in this mailing list, it might be better to use a different approach if you start everything from scratch. That will be doing all signaling handling in kamailio and use asterisk only as media server. Practically all user data is in kamailio, routing to asterisk only when needed for media services. The tutorial is more targeting existing asterisk deployments. Nevertheless, see more comments inline.
>> Here's a presentation from Astricon 2010 where I discuss multiple types of integration between Asterisk and Kamailio. The one in the tutorial
>> is, as Daniel says, focused on limited impact on an existing Asterisk installation and it's not one I recommend if you start from scratch with
>> a new architecture.
>>
>> Read it through to get a view of a couple of different approaches:
>> http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations
>>
>>
>> /O
>>
>> -----
>> Edvina SIP Masterclass in Malaga, Spain, July 2013
>> Learn more about Kamailio and SIP!
>> http://edvina.net/blog/2013/01/sipmaster-malaga-2013/
>> Register now!
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
   * http://asipto.com/u/katu *




More information about the sr-users mailing list