Stuart Henderson wrote:

> here's an update to asterisk 1.6.2.6 for anyone interested in
> testing. so far i've tested basic functions only; note that this
> has a number of significant improvements including:
>
> - additional clocking sources including a portable pthread-based timer
> - new ConfBridge application (not depending on zaptel/dahdi)
>
> diff also at http://junkpile.org/asterisk-1.6.2.6.diff
> if you try it, please let me know how you get on.

There is a problem in setup with outbound proxy (maybe related to
https://issues.asterisk.org/view.php?id=15827).

Centos 5.4 with the same 1.6.2.6 works without mentioned issue (which
is the reason why we don't use OpenBSD at the moment).

uname -a

OpenBSD node-01.my.domain 4.7 GENERIC.MP#82 amd64

debug

[Apr  7 13:49:19] NOTICE[384] manager.c: Unable to open AMI
configuration manager.conf, or configuration is invalid. Asterisk
management interface (AMI) disabled.
[Apr  7 13:49:19] NOTICE[384] cdr.c: CDR simple logging enabled.
[Apr  7 13:49:19] WARNING[384] features.c: Could not load features.conf
[Apr  7 13:49:19] NOTICE[384] loader.c: 161 modules will be loaded.
[Apr  7 13:49:19] NOTICE[384] res_smdi.c: Unable to load config
smdi.conf: SMDI disabled
[Apr  7 13:49:19] NOTICE[384] res_smdi.c: No SMDI interfaces are
available to listen on, not starting SMDI listener.
[Apr  7 13:49:19] WARNING[384] res_phoneprov.c: Unable to load users.conf
[Apr  7 13:49:19] WARNING[384] translate.c: plc_samples 160 format f
[Apr  7 13:49:19] ERROR[384] chan_unistim.c: Unable to load config unistim.conf
[Apr  7 13:49:19] NOTICE[384] chan_skinny.c: Configuring skinny from skinny.conf
[Apr  7 13:49:19] NOTICE[384] chan_skinny.c: Unable to load config
skinny.conf, Skinny disabled.
[Apr  7 13:49:19] VERBOSE[384] chan_sip.c: SIP channel loading...
[Apr  7 13:49:19] NOTICE[384] chan_sip.c: Peer '100' is now Reachable.
(29ms / 2000ms)
[Apr  7 13:49:20] ERROR[384] chan_iax2.c: Unable to load config iax.conf
[Apr  7 13:49:20] NOTICE[384] chan_agent.c: No agent configuration
found -- agent support disabled
[Apr  7 13:49:20] WARNING[384] cdr_sqlite3_custom.c: Failed to load
configuration file. Module not activated.
[Apr  7 13:49:20] WARNING[384] cdr_manager.c: Failed to load
configuration file. Module not activated.
[Apr  7 13:49:20] WARNING[384] cdr_custom.c: Failed to load
configuration file. Module not activated.
[Apr  7 13:49:20] ERROR[384] app_amd.c: Configuration file amd.conf missing.
[Apr  7 13:49:20] WARNING[384] app_followme.c: No follow me config
file (followme.conf), so no follow me
[Apr  7 13:49:20] NOTICE[384] app_queue.c: No queuerules.conf file
found, queues will not follow penalty rules
[Apr  7 13:49:20] NOTICE[384] app_queue.c: No call queueing config
file (queues.conf), so no call queues
[Apr  7 13:49:20] WARNING[384] app_minivm.c: Failed to load
configuration file. Module activated with default settings.
[Apr  7 13:49:20] ERROR[384] res_clialiases.c: res_clialiases
configuration file 'cli_aliases.conf' not found
[Apr  7 13:49:20] WARNING[384] app_festival.c: No such configuration
file festival.conf
[Apr  7 13:49:20] NOTICE[384] pbx_ael.c: Starting AEL load process.
[Apr  7 13:49:20] NOTICE[384] pbx_ael.c: File
/etc/asterisk/extensions.ael not found; AEL declining load
[Apr  7 13:49:20] ERROR[384] pbx_dundi.c: Unable to load config dundi.conf
[Apr  7 13:49:20] NOTICE[384] chan_mgcp.c: Unable to load config
mgcp.conf, MGCP disabled
[Apr  7 13:49:37] DEBUG[384] acl.c: Found IP address for this socket
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Setting SIP_TRANSPORT_UDP
with address X.X.X.X:5061
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Setting NAT on RTP to Off
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: **** Received INVITE (5) -
Command in SIP INVITE
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Setting NAT on RTP to On
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Trying to put 'SIP/2.0 401'
onto UDP socket destined for Y.Y.Y.Y:5060
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: **** Received ACK (6) -
Command in SIP ACK
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Stopping retransmission on
'[email protected]' of Response 101: Match Found
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: **** Received INVITE (5) -
Command in SIP INVITE
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Setting NAT on RTP to On
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing session-level SDP
v=0... UNSUPPORTED.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing session-level SDP
o=- 33917 33917 IN IP4 Y.Y.Y.Y... UNSUPPORTED.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing session-level SDP
s=-... UNSUPPORTED.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing session-level SDP
c=IN IP4 Y.Y.Y.Y... OK.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing session-level SDP
t=0 0... UNSUPPORTED.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing media-level
(audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing media-level
(audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing media-level
(audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing media-level
(audio) SDP a=ptime:30... OK.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Processing media-level
(audio) SDP a=sendrecv... OK.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: We're settling with these
formats: 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Checking SIP call limits for device 100
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Updating call counter for incoming call
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Joint capabilities are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Our capabilities are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** AST_CODEC_CHOOSE formats
are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: This channel will not be able
to handle video.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: build_route: Contact hop:
"100" <sip:[email protected]:5060>
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: SIP/100-00000000: New call is
still down.... Trying...
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Trying to put 'SIP/2.0 100'
onto UDP socket destined for Y.Y.Y.Y:5060
[Apr  7 13:49:37] DEBUG[384] devicestate.c: No provider found,
checking channel drivers for SIP - 100
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Checking device state for peer 100
[Apr  7 13:49:37] DEBUG[384] devicestate.c: Changing state for SIP/100
- state 1 (Not in use)
[Apr  7 13:49:37] DEBUG[384] devicestate.c: device 'SIP/100' state '1'
[Apr  7 13:49:37] DEBUG[384] pbx.c: Launching 'Dial'
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Asked to create a SIP channel
with formats: 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Setting NAT on RTP to Off
[Apr  7 13:49:37] DEBUG[384] acl.c: Found IP address for this socket
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Setting SIP_TRANSPORT_UDP
with address X.X.X.X:5061
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Joint capabilities are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Our capabilities are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** AST_CODEC_CHOOSE formats
are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: *** Our preferred formats
from the incoming channel are 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: This channel will not be able
to handle video.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable DIALEDTIME.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable ANSWEREDTIME.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable DIALEDPEERNAME.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable DIALEDPEERNUMBER.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable DIALSTATUS.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable SIPCALLID.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable SIPDOMAIN.
[Apr  7 13:49:37] DEBUG[384] channel.c: Not copying variable SIPURI.
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Outgoing Call for 999
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Updating call counter for outgoing call
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: ** Our capability: 0x4 (ulaw)
Video flag: False Text flag: False
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: -- Done with adding codecs to SDP
[Apr  7 13:49:37] DEBUG[384] channel.c: Internal timing is disabled
(option_internal_timing=0 chan->timingfd=24)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Done building SDP. Settling
with this capability: 0x4 (ulaw)
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Initializing initreq for
method INVITE - callid [email protected]
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Trying to put 'INVITE sip:'
onto UDP socket destined for X.X.X.X:5060
[Apr  7 13:49:37] WARNING[384] chan_sip.c: sip_xmit of 0x20239c000
(len 816) to X.X.X.X:5060 returned -1: Address family not supported by
protocol family
[Apr  7 13:49:37] DEBUG[384] chan_sip.c: Trying to put 'SIP/2.0 180'
onto UDP socket destined for Y.Y.Y.Y:5060
[Apr  7 13:49:38] DEBUG[384] chan_sip.c: ** SIP timers: Rescheduling
retransmission 2 to 1000 ms (t1 500 ms (Retrans id #12))
[Apr  7 13:49:38] DEBUG[384] chan_sip.c: Trying to put 'INVITE sip:'
onto UDP socket destined for X.X.X.X:5060
[Apr  7 13:49:38] WARNING[384] chan_sip.c: sip_xmit of 0x20239c000
(len 816) to X.X.X.X:5060 returned -1: Address family not supported by
protocol family
[Apr  7 13:49:39] DEBUG[384] chan_sip.c: ** SIP timers: Rescheduling
retransmission 3 to 2000 ms (t1 500 ms (Retrans id #12))
[Apr  7 13:49:39] DEBUG[384] chan_sip.c: Trying to put 'INVITE sip:'
onto UDP socket destined for X.X.X.X:5060
[Apr  7 13:49:39] WARNING[384] chan_sip.c: sip_xmit of 0x20239c000
(len 816) to X.X.X.X:5060 returned -1: Address family not supported by
protocol family
[Apr  7 13:49:41] DEBUG[384] chan_sip.c: ** SIP timers: Rescheduling
retransmission 4 to 4000 ms (t1 500 ms (Retrans id #12))
[Apr  7 13:49:41] DEBUG[384] chan_sip.c: Trying to put 'INVITE sip:'
onto UDP socket destined for X.X.X.X:5060
[Apr  7 13:49:41] WARNING[384] chan_sip.c: sip_xmit of 0x20239c000
(len 816) to X.X.X.X:5060 returned -1: Address family not supported by
protocol family
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: **** Received CANCEL (14) -
Command in SIP CANCEL
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Setting SIP_ALREADYGONE on
dialog [email protected]
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Trying to put 'SIP/2.0 487'
onto UDP socket destined for Y.Y.Y.Y:5060
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Trying to put 'SIP/2.0 200'
onto UDP socket destined for Y.Y.Y.Y:5060
[Apr  7 13:49:44] DEBUG[384] rtp.c: Channel '<unspecified>' has no
RTP, not doing anything
[Apr  7 13:49:44] DEBUG[384] channel.c: Hanging up channel 'SIP/prov-00000001'
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Hangup call
SIP/prov-00000001, SIP callid [email protected]
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Hanging up channel in state
Down (not UP)
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Acked pending invite 102
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Stopping retransmission on
'[email protected]' of Request 102: Match Found
[Apr  7 13:49:44] DEBUG[384] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Apr  7 13:49:44] DEBUG[384] pbx.c: Spawn extension (default,999,1)
exited non-zero on 'SIP/100-00000000'
[Apr  7 13:49:44] DEBUG[384] channel.c: Soft-Hanging up channel
'SIP/100-00000000'
[Apr  7 13:49:44] DEBUG[384] channel.c: Hanging up channel 'SIP/100-00000000'
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Hangup call SIP/100-00000000,
SIP callid [email protected]
[Apr  7 13:49:44] DEBUG[384] devicestate.c: No provider found,
checking channel drivers for SIP - prov
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Checking device state for peer prov
[Apr  7 13:49:44] DEBUG[384] devicestate.c: Changing state for
SIP/prov - state 1 (Not in use)
[Apr  7 13:49:44] DEBUG[384] devicestate.c: device 'SIP/prov' state '1'
[Apr  7 13:49:44] DEBUG[384] devicestate.c: No provider found,
checking channel drivers for SIP - 100
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Checking device state for peer 100
[Apr  7 13:49:44] DEBUG[384] devicestate.c: Changing state for SIP/100
- state 1 (Not in use)
[Apr  7 13:49:44] DEBUG[384] devicestate.c: device 'SIP/100' state '1'
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: **** Received ACK (6) -
Command in SIP ACK
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Stopping retransmission on
'[email protected]' of Response 102: Match Found
[Apr  7 13:49:44] DEBUG[384] chan_sip.c: Destroying SIP dialog
[email protected]
[Apr  7 13:49:50] DEBUG[384] res_musiconhold.c: Destroying MOH class 'default'
[Apr  7 13:49:50] DEBUG[384] asterisk.c: Asterisk ending (0).

sip.conf

[general]
bindaddr=X.X.X.X
bindport=5061
disallow=all
allow=ulaw

checkmwi=10
rtcachefriends=no

; Phone #1
[100]
type=friend
secret=100
nat=yes
host=dynamic
canreinvite=no
qualify=yes
mailbox=100
context=default

[openser]
insecure=port,invite
type=friend
allowsubscribe=no
context=from-trunk
host=X.X.X.X
port=5060

[prov]
type=friend
nat=no
host=Z.Z.Z.Z
port=5060
outboundproxy=X.X.X.X
outboundproxyport=5060

extensions.conf

[default]

exten => _XXX,1,Dial(SIP/${EXTEN},12,tr)
exten => _XXX,n,Hangup

exten => _3332211,1,Goto(from-trunk,_3332211,1)

exten => _999,1,Dial(SIP/${ext...@prov,120,tr)
exten => _999,n,Hangup

[from-trunk]
exten => _3332211,1,Answer()
exten => _3332211,n,Wait(1)
exten => _3332211,n,Playback(demo-thanks)
exten => _3332211,n,Dial(SIP/100,20,tr)
exten => _3332211,n,Noop(${DIALSTATUS})
exten => _3332211,n,Hangup

exten => _999,1,Dial(SIP/${ext...@prov,120,tr)
exten => _999,n,Hangup

Reply via email to