Friends,
After listening to Mark Summer’s keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other countries. He and Inveneo help groups of people to get a better understanding of how to build network, IP and voice infrastructures. One part is of course learning and managing Asterisk.

I do believe many of us wants to help his efforts, but lack the understanding and channels to reach out. I had a very brief discussion with Mark after the keynote and promised to get back to him.

My thoughts are that if anyone from these countries try to reach us, we fail to listen and help. Could be language, could be attitude or something else. We can’t expect them to have full understanding of net etiquette, the rules of Open Source project management or how to find information themselves (in a language they might not understand fully). The climate in our mailing lists and chat rooms are not always one of understanding, especially if someone copies their english language and attitude from Miami Vice

Do you have any ideas of what could be done from our community? Can we create special forums where we have a different climate, more languages and better understanding?

I also think we should copy ISOCs efforts and have a pre-astricon training/workshop for people that Inveneo locate and then invite them to Astricon, funded by grants form community or from somewhere else (since we lack an “Asterisk foundation” that could help here). I’m sure we can find resources to get them to Astricon and that we can find teachers in the community that are willing to help with this project. I would not hesitate in donating a few days myself.

We have enormous powers in our community. If we can gather a small part of that and point it towards these people, we can change the situation for many more, just by doing what we do each day – enjoy building voice solutions and sharing our knowledge.

Let’s brainstorm for a while! The floor is open.

/O

I have been using rpm version of asterisk 1.6. However, I notice the support
for gtalk is absent from rpm. I tried to compile source code and then moved
to the /usr/lib/asterisk/modules. But the modules cannot be loaded.

Anyone has successful experience.

Mine is using 1.6.2.12.

I also tried in asterisk 1.8. It works well but only the GUI is not working.

CK

Hi.

There is /doc/core-en_US.xml in asterisk 1.8 source tree. Is this file generated from documentation comments of apps/app_*.c files?

And how this file can be used? How can I convert it to pdf/html in order to use it as applications documentation?

Hi,

As we know when a user registers from some other place, his first
registration is discarded or updated to new one. i wan to know if
there is any method we can store his previous registration too. so
that we can call both of his last and current locations somewhat like
this

Dial(Sip/abc-last&Sip/abc-current)

I think this has to be done by modifying the asterisk source code. plz guide me.

Nasir Javaid
Axvoice Inc

Hi All,

for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ?

Maybe the Digium TDM410PLF card
without any daughter card
would do the job ?

Thank you very much for supporting.

Have a nice week-end,
Mike