[asterisk-users] Rining 180 and 183

[satish patel]

Thanks for reply dear

See i am going to explain my setup here

[asterisk]-----[Mediant 2000]--------E1----------[Avaya media g/w]

1) This is my setup i am useing asterisk 1.2.14 and this setup working fine but one issuse is when i call from asterisk to avaya phone i got ringback tone in my sip  phone and i can talk to party or ppls 

2) when i call from Avaya analog phone to asterisk IP phone or SIP phone i can not hear ringback tone in analog phone so how to asterisk genrate tone for avaya what is the configuration problem in my asterisk or any problem related mediant 2000 device

i can't figure out where is the problem in avaya or in mediant or asterisk 

i give you my configuration of sip

sip.conf

[115]
type=friend
context=mysip
username=115
host=dynamic
callerid=Video Phone <111>
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
subscribecontext=internal
mailbox=115 at internal

[521]
type=user
host=71.5.250.52
dtmfmode=info
secret=12345
nat=yes
context=from-trunk

[mediant]
type=friend
disallow=all
allow=alaw
context=mysip
host=dynamic
dtmfmode=info
username=mediant
secret=12345

extention.conf

[mysip]
exten => 222,1,Dial(SIP/222)
exten => 333,1,Dial(SIP/333)
exten => 555,1,Dial(SIP/555)
exten => 100,1,Dial(SIP/100)
exten => 112,1,Dial(SIP/112)
exten => 115,1,Dial(SIP/115)
exten => 611,1,Echo()

exten => _79.,1,Dial(SIP/mediant/${EXTEN:2})
exten => _79.,2,Congestion








Jared Smith <jaredsmith at jaredsmith.net> wrote: On 6/25/07, satish patel  wrote:
>                    I have confusion how to asterisk genrate tone and what
> ringing code use default 180 or 183  i have setup asterisk with mediant 2000
> with avaya

I'm assuming that you're talking SIP here... typically, if Asterisk
receives a 180 response (without SDP), it will pass that on to the end
device, which will generate the ringback tone itself.  On the other
hand, if Asterisk receives a 183 with SDP (or a 180 with SDP, which is
less common), it will try to bridge the end device with the media
specified in the SDP message.

Your best bet would be to look at the SIP messaging, specifically
looking to see if there is SDP specified.  If SDP is specified in the
183 or 180 message, then I'd try to figure out why Asterisk can't
connect to that IP address and port.  (In my experience, nine times
out of ten this is a NAT firewall problem.)

-Jared

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