[Asterisk-Users] Re: Codecs.

[Rich Adamson]

> 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.