[asterisk-users] Asterisk not sending 200 OK

Rob Schall rschall at callone.net
Wed Dec 12 08:00:49 CST 2007


Both boxes are on the outside of nats (public IPs for both). So I don't
think that would be the case. Right?

Rob


C F wrote:
> nat
>
> On 12/11/07, Rob Schall <rschall at callone.net> wrote:
>   
>> We're trying to get a SIP peer going between our asterisk box and our
>> provider. It should then ring our phone.
>>
>> The call does come in and it does execute the extension in the dial
>> plan. But the provider says they never get a 200 OK back and therefore
>> they send another INVITE and then after a few seconds drop the call.
>>
>> Here's our setup:
>>
>> sip.conf
>> [ngt-trunk]
>> type=peer
>> qualify=yes
>> port=5060
>> context=from-trunk
>> fromuser=603XXXXXXX
>> host=onecps.onvoip.net
>> registersip=no
>> username=WebSolutions
>> secret=603XXXXXXX
>> dtmfmode=inband
>> insecure=very
>>
>>
>> extensions.conf
>> [from-trunk]
>> exten => _6035467131,1,Wait(1)
>> exten => _6035467131,2,Dial(SIP/4610)
>> ;exten => _6035467131,3,Playback(ws-ivr)
>> ;exten => _6035467131,4,Hangup
>>
>>
>> The debug looks like:
>>     -- Executing [6035467131 at from-trunk:1] Wait("SIP/wsol-00820870",
>> "1") in new stack
>>     -- Executing [6035467131 at from-trunk:2] Dial("SIP/wsol-00820870",
>> "SIP/4610") in new stack
>>     -- Called 4610
>>     -- SIP/4610-00838160 is ringing
>> [Dec 11 12:18:38] NOTICE[2624]: chan_sip.c:13753 handle_request_invite:
>> Call from '' to extension '6035467131' rejected because extension not found.
>>     -- SIP/4610-00838160 answered SIP/wsol-00820870
>> Really destroying SIP dialog
>> '08c6697823d4542917eaaf607babc786 at 10.100.0.1' Method: NOTIFY
>>     -- Executing [6035467131 at from-trunk:1] Wait("SIP/wsol-00845fe0",
>> "1") in new stack
>>     -- Executing [6035467131 at from-trunk:2] Dial("SIP/wsol-00845fe0",
>> "SIP/4610") in new stack
>>     -- Called 4610
>>     -- SIP/4610-0084a310 is ringing
>> [Dec 11 12:18:49] WARNING[2624]: chan_sip.c:1939 retrans_pkt: Maximum
>> retries exceeded on transmission
>> BW1118323791112071484076745 at 63.123.133.46 for seqno 624360222 (Critical
>> Response)
>> [Dec 11 12:18:49] WARNING[2624]: chan_sip.c:1963 retrans_pkt: Hanging up
>> call BW1118323791112071484076745 at 63.123.133.46 - no reply to our
>> critical packet.
>>   == Spawn extension (from-trunk, 6035467131, 2) exited non-zero on
>> 'SIP/wsol-00820870'
>> Really destroying SIP dialog 'BW1118323791112071484076745 at 63.123.133.46'
>> Method: INVITE
>> Really destroying SIP dialog
>> '70b229a97f4c4ef260c10e6c6965c52e at 70.42.88.212' Method: INVITE
>>     -- SIP/4610-0084a310 answered SIP/wsol-00845fe0
>> [Dec 11 12:18:51] NOTICE[2624]: chan_sip.c:13753 handle_request_invite:
>> Call from '' to extension '603XXXXXXX' rejected because extension not found.
>>   == Spawn extension (from-trunk, 603XXXXXXX, 2) exited non-zero on
>> 'SIP/wsol-00845fe0'
>> Really destroying SIP dialog
>> '462aeb5a7b7f52466d86e3bc76522cd4 at 70.42.88.212' Method: BYE
>> [Dec 11 12:18:58] WARNING[2624]: chan_sip.c:1939 retrans_pkt: Maximum
>> retries exceeded on transmission
>> BW111838430111207-1062853444 at 204.11.119.46 for seqno 624363248 (Critical
>> Response)
>> Really destroying SIP dialog
>> 'BW111838430111207-1062853444 at 204.11.119.46' Method: INVITE
>> Really destroying SIP dialog
>> '14759432128ecc917770dfe13b04d62e at 70.42.88.212' Method: OPTIONS
>> Really destroying SIP dialog
>> '38f98144458b6d447752590f0018c287 at 10.100.0.1' Method: OPTIONS
>> [Dec 11 12:19:01] WARNING[2624]: chan_sip.c:1939 retrans_pkt: Maximum
>> retries exceeded on transmission
>> BW1118457001112071270417812 at 63.123.133.46 for seqno 624366883 (Critical
>> Response)
>> Really destroying SIP dialog 'BW1118457001112071270417812 at 63.123.133.46'
>> Method: INVITE
>> Really destroying SIP dialog
>> '4b2d0bc83371a9cc331c07c37ec1e5a9 at 70.42.88.212' Method: OPTIONS
>> Really destroying SIP dialog
>> '08ce62a265986f4d6229a78d74503092 at 70.42.88.212' Method: OPTIONS
>>
>>
>> _______________________________________________
>> --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
>>
>>     
>
> _______________________________________________
> --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
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071212/269c5574/attachment-0001.htm 


More information about the asterisk-users mailing list