[Asterisk-Users] Internal server error - cannot align media streams - help needed

[Arslan Saeed]

Hi,
 
I configured asterisk on redhat linux 9 box. I installed two different
ip softphones (SJPHONE and X-PRO) and got them registered with asterisk.
The call from one phone to another does get routed via asterisk, but
there is one problem coming up. As soon as call is accepted by the end
user , it is automatically disconnected with the error "cannot align
media streams". If I enable SIP debugging on asterisk, then I find the
following output 
 
 
"-- Got SIP response 500 "Internal server error (cannot align media
streams)" back from 197.7.75.129"
 
 
followed by the following debug message
 
(no NAT) to 197.7.75.129:5060
    -- SIP/2001-a513 is circuit-busy
  == Everyone is busy at this time
We're at 197.7.75.85 port 16816
Answering with preferred capability 2147483647
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
 
 
 
Below is the configuration of asterisk
 
 
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)
allow=all             ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
 
 
[2000]
 
type=soft1           ; This device takes and makes calls
username=2000         ; Username on device
secret=friend ; Password for device
host=dynamic         ; This host is not on the same IP addr every time
context=from-sip      ; Inbound calls from this host go here
mailbox=100           ; Activate the message waiting light if this
                      ; voicemailbox has messages in it
 
[2001]                ; Duplicate of 2000, except with different auth
data
 
type=soft2
username=2001
secret=friend
host=dynamic
context=from-sip
mailbox=101
 
 
 
EXTENSIONS.CONF
 
 
[general]
 
static=yes       ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
 
[bogon-calls]
 
 
 [from-sip]
 
 
 
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
 
 
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
 
 
 
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
 
 
 
 
 
Thanls
Arslan,
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