[Asterisk-Users] Problem with SIP-Phones and * audio-files

[Ernst Lehmann]

Hi All,

I am a newbie to asterisk, and here is my first problem, where I do not
know any further.

I have to grandstream BT100 connected to asterisk. Working fine, for
calling to each other, and to call via a IAX-Link to the outside.

If I try to call the initial demo from the samples.extensions.conf I
have nothing to hear.

The CLI fine reports:

    -- Executing Playback("SIP/2209-0260", "demo-abouttotry") in new
stack
    -- Playing 'demo-abouttotry' (language 'en')

after a few seconds, when I give it up....
  == Spawn extension (demo, 500, 1) exited non-zero on 'SIP/2209-0260'

When I call to the voicemail-system with extension 8500, I got also only
silence on the phone.


What can it bee ??

I tried asterisk with cvs from today (28-11-2003)

and with an older version cvs from (19-11-2003)


Thanks for any hints ....




something about the hardware:
- P4 2.8 GHz
- 1 GB RAM
- Digium E100P (but not connected at the moment)
- Digium TDM400P (but also not connected to devices at the moment)


-------------- Here my additions to the sip.conf

disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=gsm
allow=ilbc
allow=speex
allow=lpc10

; my grandstream 102
[2209]
type=friend
username=2209
secret=nosecretpasswordhere
host=dynamic
context=demo
canreinvite=yes
dtmfmode=info
qualify=yes
disallow=all
allow=g723.1
allow=ulaw
allow=alaw
allow=gsm

; my grandstream 102
[2210]
type=friend
username=2210
secret=nosecret
host=dynamic
context=demo 
canreinvite=yes
dtmfmode=info   
qualify=yes
disallow=all
allow=ulaw
allow=gsm
allow=alaw  

----------------------

in extensions.conf I only added this to lines under section [demo] for
testing the calls from gs1 -> gs2

exten => 2209,1,Dial(SIP/2209)

exten => 2210,1,Dial(SIP/2210)

-------------------------




-- 

Bye

	Ernst
---------
Ernst Lehmann             Email: lehmann at acheron.franken.de