[Asterisk-Users] Re: Codecs.
Pablo Allietti
pablo at lacnic.net
Mon Dec 19 05:26:50 MST 2005
On Sat, Dec 17, 2005 at 07:44:29AM -0600, Rich Adamson wrote:
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g1
[local]
; ignorepat => 9
include => default
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
exten => 402,1,Dial(SIP/402,20)
exten => 402,2,Hangup
[teste]
exten => s,1,Dial(SIP/402,20)
exten => s,2,Hangup
exten => 402,1,Dial(SIP/402,20)
exten => 402,2,Hangup
exten => _XXX,1,Dial(${TRUNK}/${EXTEN})
exten => _XXX,2,Voicemail(u${EXTEN})
the sip.conf is the default for asterisk i didnt touch anything in this
file only the extention number and i dont have nothing about codecs in
this file
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid="Pablo" <402>
canreinvite=no
;nat=yes
;amaflags=billing
context=teste
> > > > Hi all i have some problems with my pbx and asterisk codecs.
> > > >
> > > > if i use g711u or g711a codecs. the line never hangup. and the origin
> > > > and destination are connected until i restart my pbx or asterisk
> > > >
> > > > But if i use GSM all work fine.
> > > >
> > > > is possible to solve this problem? or use only gsm codec?
> > >
> >
> > > Yes, its possible to solve the problem.
> >
> > can you explain how?
>
> Not without you providing at least "something" to give us a clue what it
> is that you've programmed into your system.
>
> How about if you give us some clue as to which version of * you're
> using, what type of phones are associated with "origin" and "destination",
> if these are sip phones what do your sip.conf definitions look like,
> what does the appropriate sections of extensions.conf look like, and
> any other configuration pieces that might pertain to whatever it is
> that you've implemented. Your posting implies there might be more than
> one * system involved and possibly even iax trunking, etc.
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
LACNIC
More information about the asterisk-users
mailing list