[asterisk-users] Dial and connect to sip provider works, but no audio.

[Jim Lynch]

This is strange.  I upgraded from an older Asterisk at home that was 
working to the latest Tribox.  I also added a A204 board, but for some 
reason neither the Grandstream phone or a phone connected to the Linksys 
ATA has any audio either way via the Telasip connection.  Audio works OK 
between the phones, so I'm pretty sure the extension configuration is OK..

Here's my sip configs.  I added the [from-pstn] to this file because I 
didn't see it defined anywhere else.  I realize it will go away when I 
change the extensions but it wasn't working so I thought I'd try it.

I don't see much difference in configuration from when it worked and 
now, other than the missing [from-pstn] block.

Thanks for any help.

Jim.

sip_additional.conf
register=xxx.yyy at xxxxx.telasip.com

[101]
username=101
type=friend
secret=xxx
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=101 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=101 <101>

[102]
username=102
type=friend
secret=xxx
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=102 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=102 <102>

[from-pstn]
type=user
qualify=yes
insecure=very
host=xxx.telasip.com


[telasip]
username=xxx
type=friend
secret=xxx.yyy
qualify=yes
insecure=very
host=xxx.telasip.com
fromuser=xxx
fromdomain=xxx.telasip.com
dtmfmode=rfc2833
context=from-pstn

; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.

sip.conf
[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
~