asterisk users - Jan 2005 - France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo

Asterisk must have a reasonably large community here in France judging
from the number of people who came out to meet Mark. Either that or we
were ALL there :)

Something I've been waiting for, a voIP carrier on the models we are
used to (low monthly or pay as you go, web account) has just set up
their first beta test for 1 euro for the first month, 6euros if you
decide to keep it. The basic service comes with a "local" number,
i.e., similar to what you would get from France T?l?com 01, 02...
Calls are cut after two hours, but you can then call back if you need
to talk more than two hours. International rates look decent, though
not cheaper than our favorites.

The service is using a GPL SIP client, which after a few sniffs from
various Usenet denizons gave us what was needed to make the service
work with X-Lite. However, attempts to port the account over to
asterisk (which have always worked with all other providers) do not
work. I have complete session sniffs if someone is interested.

I have already began bugging them about making use with * possible.
While this is a small startup type company, they are owned by a major
telcom here, 9tel. This is the time to get a foot in the door I think.

The Windows SIP client is freely downloadable (but hard wired to work
with their service (or ip 2 ip) and they say there will be other OS
clients RealSoonNow. Since it works with X-Lite, it may well work with
linux SIP clients.

For more info: http://wengo.fr/index.php

If you are in France and learn more, please let me know. If you are
interested in sharing info, feel free to contact me off list.

On Sat, 8 Jan 2005, Wilson Pickett wrote:
> The service is using a GPL SIP client, which after a few sniffs from
> various Usenet denizons gave us what was needed to make the service
> work with X-Lite. However, attempts to port the account over to
> asterisk (which have always worked with all other providers) do not
> work. I have complete session sniffs if someone is interested.
Looks like the old problem of Asterisk not coping with SIP proxies.

Is there a real registration server on len1.host.wengo.fr instead of the 
proxy on proxy1.host.wengo.fr ?

> Is there a real registration server on len1.host.wengo.fr instead of the
> proxy on proxy1.host.wengo.fr ?
The 2 ips are different, but if I use only len1 on X-Lite it still works.

Solved. Required peer and friend.

Any chance to post a small howto with the correct server settings et al?

Thanks!
Remco


On Wed, 12 Jan 2005, Wilson Pickett wrote:
> Solved. Required peer and friend.
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On Mon, 17 Jan 2005, Wilson Pickett wrote:
>> /hax0r n00b mode on
>> Which command and parameters do I need to use to get some legible
(usable)
>> output to do the packet sniffing? I tried ethereal but it only gives me
>> loads of garbage?
>> /hax0r n00b mode off  :)
>
> Go to the Wengo forum, there is a thread in the technique section that
> gives the entire process.
One more question, if I try to dial the French telephone number I received 
I hear the tone that the number is not in use.

Is there a mistake in my asterisk config or is the French number only 
accessable from France?

> Is there a mistake in my asterisk config or is the French number only
> accessable from France?
No that happens when your server is not correctly logged in or
registered. I had that for a couple of days.

On Mon, 17 Jan 2005, Wilson Pickett wrote:
>> Can you offer any clue where I would need to look, I guess its not in
>> extensions.conf that is the problem?
> 
> 1) Are you registering with proxy1?
no, from the earlier post I understood that I shouldn't register with the 
proxy? I guess that means that I should setup 2 SIP entries, one for the 
outgoing calls (that registers with len) and another one for the incoming calls 
(that registers with proxy)?
> 2) You'll need a user or friend entry as well as the peer - at least
> that's what I did to get it working. The peer uses len1 and the friend
> uses proxy1
I tried setting it that way, but still do not get through.
sip show does show 2 connections now

Would you mind sending me the relevant bits of your sip.conf?

Thanks!!!