[asterisk-users] Rining 180 and 183

[Mojo with Horan & Company, LLC]

I think what Jared recommended, looking at the sip messaging, will help 
you here.  He means to type "sip debug" in the asterisk CLI and look for 
hints that SDP is being specified in the conversation.  If it IS being 
specified, then check into NAT/firewall issues, as he recommended also.

Mojo

satish patel wrote:
> 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
> 
>     _______________________________________________
>     --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> ------------------------------------------------------------------------
> Get your own web address. 
> <http://us.rd.yahoo.com/evt=49678/*http://smallbusiness.yahoo.com/domains/?p=BESTDEAL>
> Have a HUGE year through Yahoo! Small Business. < 
> http://us.rd.yahoo.com/evt=49678/*http://smallbusiness.yahoo.com/domains/?p=BESTDEAL> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users