[SOLVED] Dial(SIP/${EXTEN:3}:name:pass@voip_provider.com

Asterisk Asterisk Support
1

I am using Asterisk 1.2.4
I would like to make call next way:
Dial(SIP/${EXTEN:3}:name:pass@voip_provider.com
If I will try to use SJPhone - I have no problems - I can receive and make calls - same if I am using Grandstream.
If I will put in sip.conf
register => name:pass@voip_provider.com/3335
and in extensions.conf, I make extension - 3335 - I can receive calls.
But I can’t dialout.
I tried:

  1. Dial(SIP/${EXTEN:3}:name:pass@voip_provider.com
    //Also I tried all posible combinations like name:pass:${EXTEN:3}@voip_provider.com etc.
  2. In sip.conf I writed [sip_musimi-out] :
    type=peer
    secret=pass
    username=name ; Authentication user for outbound proxies
    fromuser=name ; Many SIP providers require this!
    fromdomain=my IP
    host=voip_provider.com
    usereqphone=yes ; This provider requires “;user=phone” on URI
    and tried to Dial(SIP/${EXTEN:3}@sip_musimi-out

In all cases I am receiving one and same:
– Got SIP response 500 “I’m terribly sorry, server error occurred (1/SL)” back from 212.130.58.214
– SIP/musimi.dk-5c84 is circuit-busy

I will higly appreciate any suggestions
PS
I am using musimi.dk as SIP provider.

2

Please post the /etc/asterisk/extensions.conf sectin where you are attemping your dialout and the complete CLI output for the use case.

3

in sip.conf I changed some things:

[musimi]
;;;;;;;Before when I couldn’t make connection
type=friend
secret=password
username=name
fromuser=name
fromdomain=musimi.dk
host=musimi.dk
;;;;I added next rows and everything become OK
disallow=all
allow=alaw
canreinvite=yes
context=default
dtmfmode=rfc2833
insecure=very
nat=yes

in extensions.conf:
exten => _666.,1,Dial(SIP/${EXTEN:3}@musimi)
exten => _666.,n,Hangup

This is working doesn’t metter how I will call this extension.
Meaning if I wish to call my office number through musimi I must call
666003592XXXXXXX
359 - BG code
2 Sofia code
XXXXXXX - my office number

Normaly I am using either some soft phone like SJPhone or Grandstream, which have adequate registrations in my Asterisk.
I am almost never making calls from cli.