[asterisk-bugs] [Asterisk 0014236]: handle_request_invite: Call from '101334' to extension 's' rejected because extension not found - FreeBSD sparc64

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14236 
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Reported By:                tryk
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14236
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.3-rc1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-14 04:20 CST
Last Modified:              2009-01-26 08:12 CST
====================================================================== 
Summary:                    handle_request_invite: Call from '101334' to
extension 's' rejected because extension not found - FreeBSD sparc64
Description: 
Incoming calls from external peers are rejected - looks like Asterisk is
unable to lookup the extension in dialplan on Sparc64/FreeBSD 7.0-RELEASE
build. Error message:

[Jan 14 12:09:58] NOTICE[78353]: chan_sip.c:17032 handle_request_invite:
Call from '101334' to extension 's' rejected because extension not found.

This is not a config problem, as the same configuration works perfectly on
i386 build.
====================================================================== 

---------------------------------------------------------------------- 
 (0098748) tryk (reporter) - 2009-01-26 08:12
 http://bugs.digium.com/view.php?id=14236#c98748 
---------------------------------------------------------------------- 
*****************************************************************************
*** The issue has been solved by addind IP-address as domain= in sip.conf
***
*****************************************************************************

Sorry for a stupid "bug".

[Jan 26 16:00:15] DEBUG[2332]: chan_sip.c:6062 find_call: = Found Their
Call ID: MTk3ZTM1OTU4NGM3YTFhZTJjYmVjMGQzNWQ2N2Y4NTA. Their Tag 6631e030
Our tag: as4d942390
[Jan 26 16:00:15] DEBUG[2332]: chan_sip.c:18496 handle_incoming: ****
Received REGISTER (2) - Command in SIP REGISTER
[Jan 26 16:00:15] DEBUG[2332]: chan_sip.c:18515 handle_incoming: Ignoring
SIP message because of retransmit (REGISTER Seqno 384, ours 384)
[Jan 26 16:00:15] DEBUG[2332]: chan_sip.c:2589 __sip_xmit: Trying to put
'SIP/2.0 40' onto UDP socket destined for 192.168.0.251:64445
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:6062 find_call: = Found Their
Call ID: MTk3ZTM1OTU4NGM3YTFhZTJjYmVjMGQzNWQ2N2Y4NTA. Their Tag 6631e030
Our tag: as4d942390
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:18496 handle_incoming: ****
Received REGISTER (2) - Command in SIP REGISTER
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:18515 handle_incoming: Ignoring
SIP message because of retransmit (REGISTER Seqno 384, ours 384)
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:2589 __sip_xmit: Trying to put
'SIP/2.0 40' onto UDP socket destined for 192.168.0.251:64445
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:6062 find_call: = No match Their
Call ID: MTk3ZTM1OTU4NGM3YTFhZTJjYmVjMGQzNWQ2N2Y4NTA. Their Tag 6631e030
Our tag: as4d942390
[Jan 26 16:00:19] DEBUG[2332]: acl.c:490 ast_ouraddrfor: Found IP address
for this socket
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:3915 do_setnat: Setting NAT on
RTP to Off
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:3919 do_setnat: Setting NAT on
VRTP to Off
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:5996 sip_alloc: Allocating new
SIP dialog for 99619335-8eaa-4481-9841-944329529ba6 - INVITE (With RTP)
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:18496 handle_incoming: ****
Received INVITE (5) - Command in SIP INVITE
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:3915 do_setnat: Setting NAT on
RTP to Off
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:3919 do_setnat: Setting NAT on
VRTP to Off
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:7129 process_sdp: We're settling
with these formats: 0xc (ulaw|alaw)
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:17004 handle_request_invite:
Checking SIP call limits for device 101334
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:4499 update_call_counter:
Updating call counter for incoming call
[Jan 26 16:00:19] DEBUG[2332]: devicestate.c:450
ast_devstate_changed_literal: Notification of state change to be queued on
device/channel SIP/ipshka
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:10885 get_destination: Got SIP
INVITE to non-local domain '212.109.45.71'; refusing request.
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:2589 __sip_xmit: Trying to put
'SIP/2.0 40' onto UDP socket destined for 193.28.184.13:5060
[Jan 26 16:00:19] DEBUG[2332]: devicestate.c:323 _ast_device_state: No
provider found, checking channel drivers for SIP - ipshka
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:19633 sip_devicestate: Checking
device state for peer ipshka
[Jan 26 16:00:19] DEBUG[2332]: devicestate.c:441 do_state_change: Changing
state for SIP/ipshka - state 2 (In use)
[Jan 26 16:00:19] NOTICE[2332]: chan_sip.c:17032 handle_request_invite:
Call from '101334' to extension 's' rejected because extension not found in
context 'providersin'.
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:4499 update_call_counter:
Updating call counter for incoming call
[Jan 26 16:00:19] DEBUG[2332]: devicestate.c:450
ast_devstate_changed_literal: Notification of state change to be queued on
device/channel SIP/ipshka
[Jan 26 16:00:19] DEBUG[2332]: devicestate.c:323 _ast_device_state: No
provider found, checking channel drivers for SIP - ipshka
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:19633 sip_devicestate: Checking
device state for peer ipshka
[Jan 26 16:00:19] DEBUG[2332]: devicestate.c:441 do_state_change: Changing
state for SIP/ipshka - state 1 (Not in use)
[Jan 26 16:00:19] DEBUG[2332]: app_queue.c:767 handle_statechange: Device
'SIP/ipshka' changed to state '2' (In use) but we don't care because
they're not a member of any queue.
[Jan 26 16:00:19] DEBUG[2332]: app_queue.c:767 handle_statechange: Device
'SIP/ipshka' changed to state '1' (Not in use) but we don't care because
they're not a member of any queue.
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:6062 find_call: = Found Their
Call ID: 99619335-8eaa-4481-9841-944329529ba6 Their Tag
bf60f912-52f3-425d-aac9-23bfc7c447e3 Our tag: as5d17be1b
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:18496 handle_incoming: ****
Received ACK (6) - Command in SIP ACK
[Jan 26 16:00:19] DEBUG[2332]: chan_sip.c:3066 __sip_ack: Stopping
retransmission on '99619335-8eaa-4481-9841-944329529ba6' of Response 17049:
Match Found 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-26 08:12 tryk           Note Added: 0098748                          
======================================================================