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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