[Asterisk-Users] Re: Codecs.

[Pablo Allietti]

On Mon, Dec 19, 2005 at 06:36:16AM -0600, Rich Adamson wrote:
>

ok rick i will check this directives and write you again.. thanks!!!! 
 
> > 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.
> 
> Okay, start with "show translation" to see which codecs you system
> can translate.
> 
> Then check your sip phones to see which codecs are supported. For the xten
> product (as with most sip phones), you can select which codecs to support
> and which ones are preferred.
> 
> In sip.conf you are only showing one sip phone. Are there more defined
> that you didn't paste into this email? Based on the data that you've
> provided, you only have one phone and its on extension 402. Since there
> is nothing else defined (at least based on your config files), you
> won't be able to call anyone.
> 
> You can control which codecs are used by doing something like this:
> [402]
> type=friend
> host=dynamic
> username=Pablo
> secret=teste
> callerid="Pablo" <402>
> canreinvite=no
> disallow=all
> allow=ulaw
> context=teste
> mailbox=402
> 
> [403]
> type=friend
> host=dynamic
> username=Pablo2
> secret=teste2
> callerid="Pablo" <403>
> canreinvite=no
> disallow=all
> allow=ulaw
> context=teste
> mailbox=403
> 
> Later on when you want to start playing with voicemail, you will want to
> add the statement shown above (mailbox=402).
> 
> In extensions.conf, you need entries like this:
> [teste]
> exten => 402,1,Dial(SIP/402,15)
> exten => 402,2,Voicemail(u402)
> exten => 402,102,Voicemail(b402)
> exten => 402,103,Hangup 
> 
> exten => 403,1,Dial(SIP/403,15)
> exten => 403,2,Voicemail(u403)
> exten => 403,102,Voicemail(b403)
> exten => 403,103,Hangup
> 
> With the above, extension 402 can call 403 as well as 403 can call 402.
> 
> Your entry
>  exten => s,1,Dial(SIP/402,20)
>  exten => s,2,Hangup
> does not apply to the configuration that you've shown us. The "s" extension
> is typically used for calls that arrive via Zap and Iax channels where
> "no dialed digits" are received. The "s" is not a match-all option.
> 
> We don't have any idea what you mean by "the other side". If you are 
> trying to dial from one sip phone to another on your system, then you
> need to define each phone in sip.conf as shown above, and configure
> each phone so that it properly registers with asterisk. To see what
> is registered, do a "sip show peers". If you sip phones don't show in
> that list, they aren't registered. Fix that first before moving on.
> 
> Once the above configs have been fixed and asterisk restarted, then
> watch the asterisk CLI to "see" what happens when one phone calls the
> other. If you still have problems, paste the CLI output into a posting
> for us to see. Without that, we can only guess.
> 
> Given what you have posted, I don't have a clue what you are trying to
> do with:
>  exten => _XXX,1,Dial(${TRUNK}/${EXTEN})
>  exten => _XXX,2,Voicemail(u${EXTEN})
> However, when sip extension 402 dials 403, it will match the above _XXX
> and send that call out Zap/g1 (whatever that happens to be).
> 
> If you really are working with two asterisk systems tied together with
> Zap channels, then I'd suggest modifying the above to something like
>  exten => _5XX,1,Dial(${TRUNK}/${EXTEN})
>  exten => _5XX,2,Voicemail(u${EXTEN})
> when the 4XX extensions are on one system and the 5XX extensions on the
> second system.
> 
> 
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-- 

.-

Pablo Allietti
LACNIC