[Asterisk-Users] Cisco AS5300 -- Not hearing anything

[Luciano Ramos]

Hi to all!

	I have this config,

PSTN <--> AS5300 <--> ASTERISK

	I am using the Cisco as5300 to receive incoming calls
and routing them to Asterisk for IVR.

	When I ran asterisk this is what I get when calling
the voicemail demo.

*CLI>     -- Executing Playback("SIP/-081058b8", "transfer|skip") in new
stack
    -- Executing Macro("SIP/-081058b8", "stdexten|1234|Console/dsp") in new
stack
    -- Executing Dial("SIP/-081058b8", "Console/dsp|20") in new stack
WARNING[1192437440]: File channel.c, Line 1558 (ast_request): No channel
type registered for 'Console'
NOTICE[1192437440]: File app_dial.c, Line 495 (dial_exec): Unable to create
channel of type 'Console'
  == Everyone is busy at this time
    -- Executing VoiceMail("SIP/-081058b8", "b1234") in new stack
  == Parsing '/etc/asterisk/voicemail.conf': Found
    -- Playing 'vm/1234/busy'
    -- Playing 'vm-intro'
    -- Playing 'beep'
    -- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0015
    -- User hung up
  == Parsing '/etc/asterisk/voicemail.conf': Found
  == Spawn extension (macro-stdexten, s, 102) exited non-zero on
'SIP/-081058b8' in macro 'stdexten'
  == Spawn extension (default, s, 2) exited non-zero on 'SIP/-081058b8'

But in the phone I can't hear anything, I've tested also the voicemail with
a
software sip phone and It works great. But with the cisco I hear nothing,
I'v tested codecs ulaw and alaw but the both do the same.

This is my cisco's config

dial-peer voice 20 voip
 destination-pattern 02322663910
 translate-outgoing called 20
 session protocol sipv2
 session target ipv4:200.85.96.230
 dtmf-relay cisco-rtp
 codec g711alaw
!

translation-rule 20
 Rule 0 ^02322663910 1234
!

Any ideas??


Luciano Ramos
CCNA - MCP
Jefe Depto. Internet
TelViso
02320-470300