[asterisk-users] Asterisk SIP configuration

[Anderson Luiz Brunozi]

Hi, Sherwood,

Thanks for your reply.
Here are my sip.conf and extensions.conf files.



----- begin extension.conf -----

[general]
static=yes
writeprotect=no
autofallthrough=yes


[globals]



[conference-context]
exten => s,1,Log(VERBOSE|Enter conference-context -> extension s.)
exten => s,2,Goto(conference,1)

exten => conference,1,Log(VERBOSE|Enter conference-context.)
exten => conference,2,Conference(1234/S/1)
exten => conference,3,Hangup()

----- end extension.conf -----




----- begin sip.conf -----

[general]

bindport=6060
bindaddr=0.0.0.0
srvlookup=yes

disallow=all
allow=ulaw

autocreatepeer=yes
autodomain=yes

context=conference-context

jbenable=yes
jbmaxsize=200
jbimpl=adaptive
jblog=yes


fromdomain=dmd77
realm=dmd77

register => conference at dmd77:conference at dmd77/conference

rtpkeepalive=5

insecure=invite

sipdebug=yes

nat=yes
qualify=yes

canreinvite=no



[conference]

type=friend
nat=yes
username=conference
secret=conference
canreinvite=no
host=dmd77
context=conference-context
fromdomain=dmd77


[guest]

type=friend
nat=yes
host=dynamic
canreinvite=no
context=conference-context

----- end sip.conf -----






Att,

Anderson Luiz Brunozi





-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sherwood
McGowan
Sent: Tuesday, July 29, 2008 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk SIP configuration


Anderson Luiz Brunozi wrote:
> Hello,
>  
> I'm trying to setup an instance of Asterisk to use as a SIP conference
> server only. And I'm using OpenSER as SIP Registrar/Proxy.
>  
> On asterisk's sip.conf I created a user conference at sipdomain
> <mailto:conference at sipdomain> (sipdomain == server name), that 
> registers to OpenSER when asterisk is started.
> The user conference refers to context conference-context.
>  
> And, on the extensions.conf I've defined the context called
> conference-context, with an extension named conference.
>  
> From my SIP client, I call sip:conference at sipdomain. And get a 404 Not
> found response from asterisk.
>  
>  
> On asterisk's log I see messages like:
> "Looking for conference on conference-context (domain serverIP)"
>  
> And:
> "Call from 'conference' to extension 'conference' rejected because
> extension not found."
>  
>  
> Does anyone have an ideia of why I'm getting that message?
>  
> Why does asterisk seem to be using domain == serverIP, instead of
> domain == servername? Is that correct the behavior? Or I may have 
> something missing on my configuration?
>  
> I suspect the problem is more likely be in the sip.conf file, but I
> can't see what's wrong/missing.
> I'm using app_conference. But I don't think this matters for now, 
> because my first line of the conference extension calls Log(). And, as

> I don't see my log message printed, I assume asterisk didn't even 
> start processing the commands defined for the conference extension.
>  
>  
> Thanks,
>  
> Anderson Luiz Brunozi
>  
> ----------------------------------------------------------------------
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Please send the relevant portion of your extensions.conf, as that is 
where the problem is

-- 
Sherwood McGowan
VoIP / Telecom Solutions
sherwood.mcgowan at gmail.com


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