asterisk users - Dec 2003 - Incoming callers aren't hearing ring

We just switched from three x100p's to a te410p for handling our 
incoming/outbound calls.  Everything works great, except incoming 
callers don't hear a ring while they are waiting for one of us to pick 
up.  The phones themselves ring fine, but the caller doesn't hear 
anything until someone picks up, or it transfers to voicemail.  Any 
clues as to what may be happening?

/etc/zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
defaultzone=us

/etc/asterisk/zapata.conf
[channels]
usecallerid=yes
adsi=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
rxgain=1.5
txgain=1.5
immediate=no
switchtype=national
context=incoming
signalling=pri_cpe
group=1
channel => 1-23

/etc/asterisk/extensions.conf
[incoming]
include => sip-phones
exten => _5551212,1,Dial(SIP/6710,12,tr)
exten =>
_5551212,2,Dial(SIP/6710&SIP/6711&SIP/6712&SIP/6713,20,tr)
exten => _5551212,3,Voicemail2(u6710)
exten => _5551212,4,Hangup
exten => _5551212,103,Voicemail2(b6710)
exten => _5551212,104,Hangup

You need to answer the line to place audio on the channel. So if you
place an answer line before the dials, you should get audio to route
back.

On Fri, 2003-12-26 at 23:46, Terry Wilson wrote:
> We just switched from three x100p's to a te410p for handling our 
> incoming/outbound calls.  Everything works great, except incoming 
> callers don't hear a ring while they are waiting for one of us to pick 
> up.  The phones themselves ring fine, but the caller doesn't hear 
> anything until someone picks up, or it transfers to voicemail.  Any 
> clues as to what may be happening?
> 
> /etc/zaptel.conf
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> defaultzone=us
> 
> /etc/asterisk/zapata.conf
> [channels]
> usecallerid=yes
> adsi=yes
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> rxgain=1.5
> txgain=1.5
> immediate=no
> switchtype=national
> context=incoming
> signalling=pri_cpe
> group=1
> channel => 1-23
> 
> /etc/asterisk/extensions.conf
> [incoming]
> include => sip-phones
> exten => _5551212,1,Dial(SIP/6710,12,tr)
> exten =>
_5551212,2,Dial(SIP/6710&SIP/6711&SIP/6712&SIP/6713,20,tr)
> exten => _5551212,3,Voicemail2(u6710)
> exten => _5551212,4,Hangup
> exten => _5551212,103,Voicemail2(b6710)
> exten => _5551212,104,Hangup
> 
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield <critch@basesys.com>

> you need to answer the line to place audio on the channel. So if you
> place an answer line before the dials, you should get audio to route
> back.
I just changed extensions.conf to read:

/etc/asterisk/extensions.conf
[incoming]
include => sip-phones
exten => _5551212,1,Answer
exten => _5551212,2,Dial(SIP/6710,12,tr)
exten =>
_5551212,3,Dial(SIP/6710&SIP/6711&SIP/6712&SIP/6713,20,tr)
exten => _5551212,4,Voicemail2(u6710)
exten => _5551212,5,Hangup
exten => _5551212,104,Voicemail2(b6710)
exten => _5551212,105,Hangup

and restarted asterisk for good measure. I am still not getting any ring on the
calling end.  Anything else I should check?

> Yes. Add a Ringing command.
> 
> exten => _5551212,1,Answer
> exten => _5551212,2,Ringing
> exten => _5551212,3,Dial(SIP/6710,12,tr)

Ok, extensions.conf now contains:

[incoming]
include => sip-phones
exten => _5551212,1,Answer
exten => _5551212,2,Ringing
exten => _5551212,2,Dial(SIP/6710,12,tr)

... etc. and still I am not getting a ring indicated to the caller.  It used to
work fine with the  fxo cards.  Is it possible that there is something wrong
with the way asterisk and the switch at the CLEC are communicating ring to each
other?  I appreciate the suggestions thus far, any other ideas about what could
be causing my problem?  Thanks

terry

Hi,

I move the * on a new DELL server and I get the latest version of Asterisk
with CVS. I have 3 FXO cards, X100P and the sound before was fine.
With the new version of Asterisk and on new Dell server the sound is
SOOOOOOOOO BAD!
Some suggestions are welcome.

Best regards,

Chris HARIGA