[asterisk-users] Asterisk not sending 200 OK
Rob Schall
rschall at callone.net
Tue Dec 11 12:44:17 CST 2007
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
More information about the asterisk-users
mailing list