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
